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-rw-r--r--CHANGES23
-rw-r--r--UPGRADE.txt9
-rw-r--r--apps/app_queue.c89
-rw-r--r--apps/confbridge/confbridge_manager.c2
-rw-r--r--bridges/bridge_softmix.c3
-rw-r--r--channels/chan_iax2.c8
-rw-r--r--channels/chan_sip.c139
-rw-r--r--channels/chan_skinny.c9
-rw-r--r--channels/sip/include/sip.h3
-rw-r--r--configs/samples/pjsip.conf.sample11
-rw-r--r--configs/samples/sip.conf.sample2
-rw-r--r--contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py31
-rw-r--r--contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py32
-rw-r--r--include/asterisk/network.h5
-rw-r--r--include/asterisk/res_hep.h2
-rw-r--r--include/asterisk/res_pjsip.h82
-rw-r--r--include/asterisk/res_pjsip_session.h2
-rw-r--r--include/asterisk/rtp_engine.h11
-rw-r--r--main/http.c16
-rw-r--r--main/manager.c13
-rw-r--r--main/pbx.c55
-rw-r--r--res/res_hep.c5
-rw-r--r--res/res_hep_pjsip.c12
-rw-r--r--res/res_pjsip.c151
-rw-r--r--res/res_pjsip/config_transport.c22
-rw-r--r--res/res_pjsip/pjsip_configuration.c1
-rw-r--r--res/res_pjsip/pjsip_message_ip_updater.c83
-rw-r--r--res/res_pjsip_nat.c43
-rw-r--r--res/res_pjsip_pubsub.c46
-rw-r--r--res/res_pjsip_refer.c7
-rw-r--r--res/res_pjsip_sdp_rtp.c89
-rw-r--r--res/res_pjsip_session.c39
-rw-r--r--res/res_pjsip_t38.c5
-rw-r--r--res/res_rtp_asterisk.c369
34 files changed, 1080 insertions, 339 deletions
diff --git a/CHANGES b/CHANGES
index ad1ff237b..08c9185b6 100644
--- a/CHANGES
+++ b/CHANGES
@@ -41,6 +41,22 @@ app_voicemail
* Added 'fromstring' field to the voicemail boxes. If set, it will override
the global 'fromstring' field on a per-mailbox basis.
+res_pjsip
+------------------
+ * A new transport parameter 'symmetric_transport' has been added.
+ When a request from a dynamic contact comes in on a transport with this
+ option set to 'yes', the transport name will be saved and used for
+ subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's
+ saved as a contact uri parameter named 'x-ast-txp' and will display with
+ the contact uri in CLI, AMI, and ARI output. On the outgoing request,
+ if a transport wasn't explicitly set on the endpoint AND the request URI
+ is not a hostname, the saved transport will be used and the 'x-ast-txp'
+ parameter stripped from the outgoing packet. To facilitate recreation of
+ subscriptions on asterisk restart, a new column 'contact_uri' needed to be
+ added to the ps_subcsription_persistence table. Since new columns were
+ added to both transport and subscription_persistence, an alembic upgrade
+ should be run to bring the database tables up to date.
+
res_pjsip_transport_websocket
------------------
* Removed non-secure websocket support. Firefox and Chrome have not allowed
@@ -63,6 +79,13 @@ res_pjsip_endpoint_identifier_ip
appropriate, as it now matches inbound requests on more than just IP
address.
+res_rtp_asterisk
+-----------------
+ * The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
+ Data and Control Packets on a Single Port." So far, the only channel driver
+ that supports this feature is chan_pjsip. You can set "rtcp_mux = yes" on
+ a PJSIP endpoint in pjsip.conf to enable the feature.
+
------------------------------------------------------------------------------
--- Functionality changes from Asterisk 13.13.0 to Asterisk 13.14.0 ----------
------------------------------------------------------------------------------
diff --git a/UPGRADE.txt b/UPGRADE.txt
index 4f40b2b9d..63a1885d5 100644
--- a/UPGRADE.txt
+++ b/UPGRADE.txt
@@ -21,6 +21,15 @@
=== UPGRADE-12.txt -- Upgrade info for 11 to 12
===========================================================
+From 13.14.0 to 13.15.0:
+
+res_rtp_asterisk:
+ - The RTP layer of Asterisk now has support for RFC 5761: "Multiplexing RTP
+ Data and Control Packets on a Single Port." For the PJSIP channel driver,
+ chan_pjsip, you can set "rtcp_mux = yes" on a PJSIP endpoint in pjsip.conf
+ to enable the feature. For chan_sip you can set "rtcp_mux = yes" either
+ globally or on a per-peer basis in sip.conf.
+
From 13.8.0 to 13.9.0:
res_parking:
diff --git a/apps/app_queue.c b/apps/app_queue.c
index f7cee6a76..ddb62d2e0 100644
--- a/apps/app_queue.c
+++ b/apps/app_queue.c
@@ -5884,6 +5884,7 @@ static void handle_bridge_enter(void *userdata, struct stasis_subscription *sub,
{
struct queue_stasis_data *queue_data = userdata;
struct ast_bridge_blob *enter_blob = stasis_message_data(msg);
+ SCOPED_AO2LOCK(lock, queue_data);
if (queue_data->dying) {
return;
@@ -5902,6 +5903,67 @@ static void handle_bridge_enter(void *userdata, struct stasis_subscription *sub,
}
/*!
+ * \internal
+ * \brief Handle a stasis bridge leave event.
+ *
+ * We track this event to determine if the caller has left the bridge
+ * as the result of a redirect. Transfers and hangups are handled in
+ * separate functions.
+ *
+ * \param userdata Data pertaining to the particular call in the queue.
+ * \param sub The stasis subscription on which the message occurred.
+ * \param msg The stasis message for the bridge leave event
+ */
+static void handle_bridge_left(void *userdata, struct stasis_subscription *sub,
+ struct stasis_message *msg)
+{
+ struct queue_stasis_data *queue_data = userdata;
+ struct ast_bridge_blob *left_blob = stasis_message_data(msg);
+ struct ast_channel_snapshot *caller_snapshot, *member_snapshot;
+
+ ao2_lock(queue_data);
+
+ if (queue_data->dying) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
+ if (ast_strlen_zero(queue_data->bridge_uniqueid)) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
+ /* Correct channel, correct bridge? */
+ if (strcmp(left_blob->channel->uniqueid, queue_data->caller_uniqueid)
+ || strcmp(left_blob->bridge->uniqueid, queue_data->bridge_uniqueid)) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
+ caller_snapshot = ast_channel_snapshot_get_latest(queue_data->caller_uniqueid);
+ member_snapshot = ast_channel_snapshot_get_latest(queue_data->member_uniqueid);
+
+ ao2_unlock(queue_data);
+
+ ast_debug(3, "Detected redirect of queue caller channel %s\n",
+ caller_snapshot->name);
+
+ ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
+ "COMPLETECALLER", "%ld|%ld|%d",
+ (long) (queue_data->starttime - queue_data->holdstart),
+ (long) (time(NULL) - queue_data->starttime), queue_data->caller_pos);
+
+ send_agent_complete(queue_data->queue->name, caller_snapshot, member_snapshot, queue_data->member,
+ queue_data->holdstart, queue_data->starttime, CALLER);
+ update_queue(queue_data->queue, queue_data->member, queue_data->callcompletedinsl,
+ time(NULL) - queue_data->starttime);
+ remove_stasis_subscriptions(queue_data);
+
+ ao2_cleanup(member_snapshot);
+ ao2_cleanup(caller_snapshot);
+}
+
+/*!
* \brief Handle a blind transfer event
*
* This event is important in order to be able to log the end of the
@@ -5922,16 +5984,17 @@ static void handle_blind_transfer(void *userdata, struct stasis_subscription *su
RAII_VAR(struct ast_channel_snapshot *, caller_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, member_snapshot, NULL, ao2_cleanup);
- if (queue_data->dying) {
- return;
- }
-
if (transfer_msg->result != AST_BRIDGE_TRANSFER_SUCCESS) {
return;
}
ao2_lock(queue_data);
+ if (queue_data->dying) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
if (ast_strlen_zero(queue_data->bridge_uniqueid) ||
strcmp(queue_data->bridge_uniqueid, transfer_msg->bridge->uniqueid)) {
ao2_unlock(queue_data);
@@ -5979,10 +6042,6 @@ static void handle_attended_transfer(void *userdata, struct stasis_subscription
RAII_VAR(struct ast_channel_snapshot *, caller_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, member_snapshot, NULL, ao2_cleanup);
- if (queue_data->dying) {
- return;
- }
-
if (atxfer_msg->result != AST_BRIDGE_TRANSFER_SUCCESS ||
atxfer_msg->dest_type == AST_ATTENDED_TRANSFER_DEST_THREEWAY) {
return;
@@ -5990,6 +6049,11 @@ static void handle_attended_transfer(void *userdata, struct stasis_subscription
ao2_lock(queue_data);
+ if (queue_data->dying) {
+ ao2_unlock(queue_data);
+ return;
+ }
+
if (ast_strlen_zero(queue_data->bridge_uniqueid)) {
ao2_unlock(queue_data);
return;
@@ -6173,12 +6237,13 @@ static void handle_hangup(void *userdata, struct stasis_subscription *sub,
RAII_VAR(struct ast_channel *, chan, NULL, ao2_cleanup);
enum agent_complete_reason reason;
+ ao2_lock(queue_data);
+
if (queue_data->dying) {
+ ao2_unlock(queue_data);
return;
}
- ao2_lock(queue_data);
-
if (!strcmp(channel_blob->snapshot->uniqueid, queue_data->caller_uniqueid)) {
reason = CALLER;
} else if (!strcmp(channel_blob->snapshot->uniqueid, queue_data->member_uniqueid)) {
@@ -6207,7 +6272,7 @@ static void handle_hangup(void *userdata, struct stasis_subscription *sub,
ast_debug(3, "Detected hangup of queue %s channel %s\n", reason == CALLER ? "caller" : "member",
channel_blob->snapshot->name);
- ast_queue_log(queue_data->queue->name, queue_data->caller_uniqueid, queue_data->member->membername,
+ ast_queue_log(queue_data->queue->name, caller_snapshot->uniqueid, queue_data->member->membername,
reason == CALLER ? "COMPLETECALLER" : "COMPLETEAGENT", "%ld|%ld|%d",
(long) (queue_data->starttime - queue_data->holdstart),
(long) (time(NULL) - queue_data->starttime), queue_data->caller_pos);
@@ -6268,6 +6333,8 @@ static int setup_stasis_subs(struct queue_ent *qe, struct ast_channel *peer, str
stasis_message_router_add(queue_data->bridge_router, ast_channel_entered_bridge_type(),
handle_bridge_enter, queue_data);
+ stasis_message_router_add(queue_data->bridge_router, ast_channel_left_bridge_type(),
+ handle_bridge_left, queue_data);
stasis_message_router_add(queue_data->bridge_router, ast_blind_transfer_type(),
handle_blind_transfer, queue_data);
stasis_message_router_add(queue_data->bridge_router, ast_attended_transfer_type(),
diff --git a/apps/confbridge/confbridge_manager.c b/apps/confbridge/confbridge_manager.c
index eb1b58e15..bca854ed9 100644
--- a/apps/confbridge/confbridge_manager.c
+++ b/apps/confbridge/confbridge_manager.c
@@ -191,7 +191,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
</managerEvent>
<managerEvent language="en_US" name="ConfbridgeTalking">
<managerEventInstance class="EVENT_FLAG_CALL">
- <synopsis>Raised when a confbridge participant unmutes.</synopsis>
+ <synopsis>Raised when a confbridge participant begins or ends talking.</synopsis>
<syntax>
<parameter name="Conference">
<para>The name of the Confbridge conference.</para>
diff --git a/bridges/bridge_softmix.c b/bridges/bridge_softmix.c
index 436fab7af..486330af0 100644
--- a/bridges/bridge_softmix.c
+++ b/bridges/bridge_softmix.c
@@ -306,7 +306,8 @@ static void softmix_process_write_audio(struct softmix_translate_helper *trans_h
if (entry->trans_pvt && !entry->out_frame) {
entry->out_frame = ast_translate(entry->trans_pvt, &sc->write_frame, 0);
}
- if (entry->out_frame && (entry->out_frame->datalen < MAX_DATALEN)) {
+ if (entry->out_frame && entry->out_frame->frametype == AST_FRAME_VOICE
+ && entry->out_frame->datalen < MAX_DATALEN) {
ao2_replace(sc->write_frame.subclass.format, entry->out_frame->subclass.format);
memcpy(sc->final_buf, entry->out_frame->data.ptr, entry->out_frame->datalen);
sc->write_frame.datalen = entry->out_frame->datalen;
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index db5e4fa4d..e8c47ce9f 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -12928,7 +12928,13 @@ static struct iax2_peer *build_peer(const char *name, struct ast_variable *v, st
/* Non-dynamic. Make sure we become that way if we're not */
AST_SCHED_DEL(sched, peer->expire);
ast_clear_flag64(peer, IAX_DYNAMIC);
- peer->addr.ss.ss_family = AST_AF_UNSPEC;
+ if (peer->dnsmgr) {
+ // Make sure we refresh dnsmgr if we're using it
+ ast_dnsmgr_refresh(peer->dnsmgr);
+ } else {
+ // Or just invalidate the address
+ peer->addr.ss.ss_family = AST_AF_UNSPEC;
+ }
if (ast_dnsmgr_lookup(v->value, &peer->addr, &peer->dnsmgr, srvlookup ? "_iax._udp" : NULL)) {
return peer_unref(peer);
}
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 85796a073..a67ad634c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1211,6 +1211,7 @@ static int process_sdp_o(const char *o, struct sip_pvt *p);
static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
static int process_sdp_a_sendonly(const char *a, int *sendonly);
static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
+static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested);
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
@@ -6008,7 +6009,7 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
ast_rtp_instance_set_qos(dialog->vrtp, global_tos_video, global_cos_video, "SIP VIDEO");
}
@@ -6028,14 +6029,14 @@ static int dialog_initialize_rtp(struct sip_pvt *dialog)
/* Do not timeout text as its not constant*/
ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
}
ast_rtp_instance_set_timeout(dialog->rtp, dialog->rtptimeout);
ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
ast_rtp_instance_set_keepalive(dialog->rtp, dialog->rtpkeepalive);
- ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
@@ -7754,6 +7755,15 @@ static int interpret_t38_parameters(struct sip_pvt *p, const struct ast_control_
return res;
}
+enum sip_media_fds {
+ SIP_AUDIO_RTP_FD,
+ SIP_AUDIO_RTCP_FD,
+ SIP_VIDEO_RTP_FD,
+ SIP_VIDEO_RTCP_FD,
+ SIP_TEXT_RTP_FD,
+ SIP_UDPTL_FD,
+};
+
/*!
* \internal
* \brief Create and initialize UDPTL for the specified dialog
@@ -7782,7 +7792,7 @@ static int initialize_udptl(struct sip_pvt *p)
/* T38 can be supported by this dialog, create it and set the derived properties */
if ((p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr))) {
if (p->owner) {
- ast_channel_set_fd(p->owner, 5, ast_udptl_fd(p->udptl));
+ ast_channel_set_fd(p->owner, SIP_UDPTL_FD, ast_udptl_fd(p->udptl));
}
ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
@@ -8208,20 +8218,28 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
* UDPTL is created as needed in the lifetime of a dialog, its file
* descriptor is set in initialize_udptl */
if (i->rtp) {
- ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
- ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTP_FD, ast_rtp_instance_fd(i->rtp, 0));
+ if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, -1);
+ } else {
+ ast_channel_set_fd(tmp, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(i->rtp, 1));
+ }
ast_rtp_instance_set_write_format(i->rtp, fmt);
ast_rtp_instance_set_read_format(i->rtp, fmt);
}
if (needvideo && i->vrtp) {
- ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
- ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTP_FD, ast_rtp_instance_fd(i->vrtp, 0));
+ if (ast_test_flag(&i->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, -1);
+ } else {
+ ast_channel_set_fd(tmp, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(i->vrtp, 1));
+ }
}
if (needtext && i->trtp) {
- ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
+ ast_channel_set_fd(tmp, SIP_TEXT_RTP_FD, ast_rtp_instance_fd(i->trtp, 0));
}
if (i->udptl) {
- ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
+ ast_channel_set_fd(tmp, SIP_UDPTL_FD, ast_udptl_fd(i->udptl));
}
if (state == AST_STATE_RING) {
@@ -10090,6 +10108,42 @@ static int has_media_stream(struct sip_pvt *p, enum media_type m)
return 0;
}
+static void configure_rtcp(struct sip_pvt *p, struct ast_rtp_instance *instance, int which, int remote_rtcp_mux)
+{
+ int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+ int fd = -1;
+
+ if (local_rtcp_mux && remote_rtcp_mux) {
+ ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
+ } else {
+ ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
+ fd = ast_rtp_instance_fd(instance, 1);
+ }
+
+ if (p->owner) {
+ ast_channel_set_fd(p->owner, which, fd);
+ }
+}
+
+static void set_ice_components(struct sip_pvt *p, struct ast_rtp_instance *instance, int remote_rtcp_mux)
+{
+ struct ast_rtp_engine_ice *ice;
+ int local_rtcp_mux = ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX);
+
+ ice = ast_rtp_instance_get_ice(instance);
+ if (!ice) {
+ return;
+ }
+
+ if (local_rtcp_mux && remote_rtcp_mux) {
+ /* We both support RTCP mux. Only one ICE component necessary */
+ ice->change_components(instance, 1);
+ } else {
+ /* They either don't support RTCP mux or we don't know if they do yet. */
+ ice->change_components(instance, 2);
+ }
+}
+
/*! \brief Process SIP SDP offer, select formats and activate media channels
If offer is rejected, we will not change any properties of the call
Return 0 on success, a negative value on errors.
@@ -10148,6 +10202,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int secure_audio = FALSE;
int secure_video = FALSE;
+ /* RTCP Multiplexing */
+ int remote_rtcp_mux_audio = FALSE;
+ int remote_rtcp_mux_video = FALSE;
+
/* Others */
int sendonly = -1;
unsigned int numberofports;
@@ -10674,6 +10732,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
processed = TRUE;
} else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
processed = TRUE;
+ } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_audio)) {
+ processed = TRUE;
}
}
/* Video specific scanning */
@@ -10691,6 +10751,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
processed = TRUE;
} else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
processed = TRUE;
+ } else if (process_sdp_a_rtcp_mux(value, p, &remote_rtcp_mux_video)) {
+ processed = TRUE;
}
}
/* Text (T.140) specific scanning */
@@ -10855,6 +10917,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (sa && portno > 0) {
/* Start ICE negotiation here, only when it is response, and setting that we are conrolling agent,
as we are offerer */
+ set_ice_components(p, p->rtp, remote_rtcp_mux_audio);
if (req->method == SIP_RESPONSE) {
start_ice(p->rtp, 1);
}
@@ -10868,11 +10931,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
/* Ensure RTCP is enabled since it may be inactive
if we're coming back from a T.38 session */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
- /* Ensure audio RTCP reads are enabled */
- if (p->owner) {
- ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
- }
+ configure_rtcp(p, p->rtp, SIP_AUDIO_RTCP_FD, remote_rtcp_mux_audio);
if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
ast_clear_flag(&p->flags[0], SIP_DTMF);
@@ -10895,10 +10954,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Prevent audio RTCP reads */
if (p->owner) {
- ast_channel_set_fd(p->owner, 1, -1);
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
}
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
} else {
ast_rtp_instance_stop(p->rtp);
if (debug)
@@ -10909,6 +10968,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Setup video address and port */
if (p->vrtp) {
if (vsa && vportno > 0) {
+ set_ice_components(p, p->vrtp, remote_rtcp_mux_video);
start_ice(p->vrtp, (req->method != SIP_RESPONSE) ? 0 : 1);
ast_sockaddr_set_port(vsa, vportno);
ast_rtp_instance_set_remote_address(p->vrtp, vsa);
@@ -10917,6 +10977,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_sockaddr_stringify(vsa));
}
ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
+ configure_rtcp(p, p->vrtp, SIP_VIDEO_RTCP_FD, remote_rtcp_mux_video);
} else {
ast_rtp_instance_stop(p->vrtp);
if (debug)
@@ -11263,6 +11324,18 @@ static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_in
return found;
}
+static int process_sdp_a_rtcp_mux(const char *a, struct sip_pvt *p, int *requested)
+{
+ int found = FALSE;
+
+ if (!strncasecmp(a, "rtcp-mux", 8)) {
+ *requested = TRUE;
+ found = TRUE;
+ }
+
+ return found;
+}
+
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_dtls *dtls;
@@ -13617,6 +13690,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
add_dtls_to_sdp(p->rtp, &a_audio);
}
+
+ /* If we've got rtcp-mux enabled, just unconditionally offer it in all SDPs */
+ if (ast_test_flag(&p->flags[2], SIP_PAGE3_RTCP_MUX)) {
+ ast_str_append(&a_audio, 0, "a=rtcp-mux\r\n");
+ ast_str_append(&a_video, 0, "a=rtcp-mux\r\n");
+ }
}
if (add_t38) {
@@ -13984,18 +14063,18 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int old
if (p->rtp) {
if (t38version) {
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
if (p->owner) {
/* Prevent audio RTCP reads */
- ast_channel_set_fd(p->owner, 1, -1);
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, -1);
}
} else if (ast_sockaddr_isnull(&p->redirip)) {
/* Enable RTCP since it will be inactive if we're coming back
* with this reinvite */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
if (p->owner) {
/* Enable audio RTCP reads */
- ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
+ ast_channel_set_fd(p->owner, SIP_AUDIO_RTCP_FD, ast_rtp_instance_fd(p->rtp, 1));
}
}
}
@@ -20963,6 +21042,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
ast_cli(fd, " Parkinglot : %s\n", peer->parkinglot);
ast_cli(fd, " Use Reason : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
ast_cli(fd, " Encryption : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)));
+ ast_cli(fd, " RTCP Mux : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX)));
ast_cli(fd, "\n");
peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer ptr");
} else if (peer && type == 1) { /* manager listing */
@@ -21033,6 +21113,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
astman_append(s, "SIP-Encryption: %s\r\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP) ? "Y" : "N");
+ astman_append(s, "SIP-RTCP-Mux: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_RTCP_MUX) ? "Y" : "N");
/* - is enumerated */
astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
@@ -21657,6 +21738,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " MOH Interpret: %s\n", default_mohinterpret);
ast_cli(a->fd, " MOH Suggest: %s\n", default_mohsuggest);
ast_cli(a->fd, " Voice Mail Extension: %s\n", default_vmexten);
+ ast_cli(a->fd, " RTCP Multiplexing: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[2], SIP_PAGE3_RTCP_MUX)));
if (realtimepeers || realtimeregs) {
@@ -30710,6 +30792,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask
} else if (!strcasecmp(v->name, "buggymwi")) {
ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
+ } else if (!strcasecmp(v->name, "rtcp_mux")) {
+ ast_set_flag(&mask[2], SIP_PAGE3_RTCP_MUX);
+ ast_set2_flag(&flags[2], ast_true(v->value), SIP_PAGE3_RTCP_MUX);
} else
res = 0;
@@ -33311,9 +33396,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
if (p->rtp) {
/* Prevent audio RTCP reads */
- ast_channel_set_fd(chan, 1, -1);
+ ast_channel_set_fd(chan, SIP_AUDIO_RTCP_FD, -1);
/* Silence RTCP while audio RTP is inactive */
- ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
}
} else if (!ast_sockaddr_isnull(&p->redirip)) {
memset(&p->redirip, 0, sizeof(p->redirip));
@@ -33325,9 +33410,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
if (p->vrtp) {
/* Prevent video RTCP reads */
- ast_channel_set_fd(chan, 3, -1);
+ ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, -1);
/* Silence RTCP while video RTP is inactive */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 0);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_DISABLED);
}
} else if (!ast_sockaddr_isnull(&p->vredirip)) {
memset(&p->vredirip, 0, sizeof(p->vredirip));
@@ -33336,9 +33421,9 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *i
if (p->vrtp) {
/* Enable RTCP since it will be inactive if we're coming back
* from a reinvite */
- ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 1);
+ ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD);
/* Enable video RTCP reads */
- ast_channel_set_fd(chan, 3, ast_rtp_instance_fd(p->vrtp, 1));
+ ast_channel_set_fd(chan, SIP_VIDEO_RTCP_FD, ast_rtp_instance_fd(p->vrtp, 1));
}
}
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 9cde77540..9c84eec30 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -7642,7 +7642,6 @@ static void *accept_thread(void *ignore)
struct sockaddr_in sin;
socklen_t sinlen;
struct skinnysession *s;
- struct protoent *p;
int arg = 1;
for (;;) {
@@ -7659,12 +7658,10 @@ static void *accept_thread(void *ignore)
continue;
}
- p = getprotobyname("tcp");
- if(p) {
- if( setsockopt(as, p->p_proto, TCP_NODELAY, (char *)&arg, sizeof(arg) ) < 0 ) {
- ast_log(LOG_WARNING, "Failed to set Skinny tcp connection to TCP_NODELAY mode: %s\n", strerror(errno));
- }
+ if (setsockopt(as, IPPROTO_TCP, TCP_NODELAY, (char *) &arg, sizeof(arg)) < 0) {
+ ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on Skinny TCP connection: %s\n", strerror(errno));
}
+
if (!(s = ast_calloc(1, sizeof(struct skinnysession)))) {
close(as);
ast_atomic_fetchadd_int(&unauth_sessions, -1);
diff --git a/channels/sip/include/sip.h b/channels/sip/include/sip.h
index 44c8ddf35..850370cd7 100644
--- a/channels/sip/include/sip.h
+++ b/channels/sip/include/sip.h
@@ -384,11 +384,12 @@
#define SIP_PAGE3_IGNORE_PREFCAPS (1 << 7) /*!< DP: Ignore prefcaps when setting up an outgoing call leg */
#define SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL (1 << 8) /*!< DGP: Stop telling the peer to start music on hold */
#define SIP_PAGE3_FORCE_AVP (1 << 9) /*!< DGP: Force 'RTP/AVP' for all streams, even DTLS */
+#define SIP_PAGE3_RTCP_MUX (1 << 10) /*!< DGP: Attempt to negotiate RFC 5761 RTCP multiplexing */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32 | SIP_PAGE3_NAT_AUTO_RPORT | SIP_PAGE3_NAT_AUTO_COMEDIA | \
SIP_PAGE3_DIRECT_MEDIA_OUTGOING | SIP_PAGE3_USE_AVPF | SIP_PAGE3_ICE_SUPPORT | SIP_PAGE3_IGNORE_PREFCAPS | \
- SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP)
+ SIP_PAGE3_DISCARD_REMOTE_HOLD_RETRIEVAL | SIP_PAGE3_FORCE_AVP | SIP_PAGE3_RTCP_MUX)
#define CHECK_AUTH_BUF_INITLEN 256
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index f66161329..82da311a0 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -841,6 +841,17 @@
; this option is set to 'no' (the default) changes to the
; particular transport will be ignored. If set to 'yes',
; changes (if any) will be applied.
+;symmetric_transport=no ; When a request from a dynamic contact comes in on a
+ ; transport with this option set to 'yes', the transport
+ ; name will be saved and used for subsequent outgoing
+ ; requests like OPTIONS, NOTIFY and INVITE. It's saved
+ ; as a contact uri parameter named 'x-ast-txp' and will
+ ; display with the contact uri in CLI, AMI, and ARI
+ ; output. On the outgoing request, if a transport
+ ; wasn't explicitly set on the endpoint AND the request
+ ; URI is not a hostname, the saved transport will be
+ ; used and the 'x-ast-txp' parameter stripped from the
+ ; outgoing packet.
;==========================AOR SECTION OPTIONS=========================
;[aor]
diff --git a/configs/samples/sip.conf.sample b/configs/samples/sip.conf.sample
index c5ffdcccd..2ef997036 100644
--- a/configs/samples/sip.conf.sample
+++ b/configs/samples/sip.conf.sample
@@ -1063,6 +1063,8 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; option may be specified at the global or peer scope.
;force_avp=yes ; Force 'RTP/AVP', 'RTP/AVPF', 'RTP/SAVP', and 'RTP/SAVPF' to be used for
; media streams when appropriate, even if a DTLS stream is present.
+;rtcp_mux=yes ; Enable support for RFC 5761 RTCP multiplexing which is required for
+ ; WebRTC support
; ---------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,
; please read https://wiki.asterisk.org/wiki/display/AST/Realtime+Database+Configuration
diff --git a/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py b/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py
new file mode 100644
index 000000000..8b0214a17
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/15db7b91a97a_add_rtcp_mux.py
@@ -0,0 +1,31 @@
+"""empty message
+
+Revision ID: 15db7b91a97a
+Revises: 465e70e8c337
+Create Date: 2017-03-08 16:56:38.108162
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = '15db7b91a97a'
+down_revision = '465e70e8c337'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+ ############################# Enums ##############################
+
+ # yesno_values have already been created, so use postgres enum object
+ # type to get around "already created" issue - works okay with mysql
+ yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+ op.add_column('ps_endpoints', sa.Column('rtcp_mux', yesno_values))
+
+
+def downgrade():
+ op.drop_column('ps_endpoints', 'rtcp_mux')
diff --git a/contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py b/contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py
new file mode 100644
index 000000000..51b5066f5
--- /dev/null
+++ b/contrib/ast-db-manage/config/versions/f638dbe2eb23_symmetric_transport.py
@@ -0,0 +1,32 @@
+"""symmetric_transport
+
+Revision ID: f638dbe2eb23
+Revises: 15db7b91a97a
+Create Date: 2017-03-09 09:38:59.513479
+
+"""
+
+# revision identifiers, used by Alembic.
+revision = 'f638dbe2eb23'
+down_revision = '15db7b91a97a'
+
+from alembic import op
+import sqlalchemy as sa
+from sqlalchemy.dialects.postgresql import ENUM
+
+YESNO_NAME = 'yesno_values'
+YESNO_VALUES = ['yes', 'no']
+
+def upgrade():
+ ############################# Enums ##############################
+
+ # yesno_values have already been created, so use postgres enum object
+ # type to get around "already created" issue - works okay with mysql
+ yesno_values = ENUM(*YESNO_VALUES, name=YESNO_NAME, create_type=False)
+
+ op.add_column('ps_transports', sa.Column('symmetric_transport', yesno_values))
+ op.add_column('ps_subscription_persistence', sa.Column('contact_uri', sa.String(256)))
+
+def downgrade():
+ op.drop_column('ps_subscription_persistence', 'contact_uri')
+ op.drop_column('ps_transports', 'symmetric_transport')
diff --git a/include/asterisk/network.h b/include/asterisk/network.h
index 3371e5895..5216f4c61 100644
--- a/include/asterisk/network.h
+++ b/include/asterisk/network.h
@@ -86,6 +86,11 @@ const char *ast_inet_ntoa(struct in_addr ia);
#endif
#define inet_ntoa __dont__use__inet_ntoa__use__ast_inet_ntoa__instead__
+#ifdef getprotobyname
+#undef getprotobyname
+#endif
+#define getprotobyname __getprotobyname_is_not_threadsafe__do_not_use__
+
/*! \brief Compares the source address and port of two sockaddr_in */
static force_inline int inaddrcmp(const struct sockaddr_in *sin1, const struct sockaddr_in *sin2)
{
diff --git a/include/asterisk/res_hep.h b/include/asterisk/res_hep.h
index cfd213ad7..dba86e88b 100644
--- a/include/asterisk/res_hep.h
+++ b/include/asterisk/res_hep.h
@@ -72,6 +72,8 @@ struct hepv3_capture_info {
size_t len;
/*! If non-zero, the payload accompanying this capture info will be compressed */
unsigned int zipped:1;
+ /*! The IPPROTO_* protocol where we captured the packet */
+ int protocol_id;
};
/*!
diff --git a/include/asterisk/res_pjsip.h b/include/asterisk/res_pjsip.h
index d8e172fc5..05a3eea44 100644
--- a/include/asterisk/res_pjsip.h
+++ b/include/asterisk/res_pjsip.h
@@ -194,6 +194,8 @@ struct ast_sip_transport {
int write_timeout;
/*! Allow reload */
int allow_reload;
+ /*! Automatically send requests out the same transport requests have come in on */
+ int symmetric_transport;
};
#define SIP_SORCERY_DOMAIN_ALIAS_TYPE "domain_alias"
@@ -755,8 +757,14 @@ struct ast_sip_endpoint {
char *contact_user;
/*! Do we allow an asymmetric RTP codec? */
unsigned int asymmetric_rtp_codec;
+ /*! Use RTCP-MUX */
+ unsigned int rtcp_mux;
};
+/*! URI parameter for symmetric transport */
+#define AST_SIP_X_AST_TXP "x-ast-txp"
+#define AST_SIP_X_AST_TXP_LEN 9
+
/*!
* \brief Initialize an auth vector with the configured values.
*
@@ -1700,6 +1708,26 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
/*!
* \brief General purpose method for creating an rdata structure using specific information
+ * \since 13.15.0
+ *
+ * \param rdata[out] The rdata structure that will be populated
+ * \param packet A SIP message
+ * \param src_name The source IP address of the message
+ * \param src_port The source port of the message
+ * \param transport_type The type of transport the message was received on
+ * \param local_name The local IP address the message was received on
+ * \param local_port The local port the message was received on
+ * \param contact_uri The contact URI of the message
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ */
+int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet,
+ const char *src_name, int src_port, char *transport_type, const char *local_name,
+ int local_port, const char *contact_uri);
+
+/*!
+ * \brief General purpose method for creating an rdata structure using specific information
*
* \param rdata[out] The rdata structure that will be populated
* \param packet A SIP message
@@ -1712,8 +1740,8 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
* \retval 0 success
* \retval -1 failure
*/
-int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port, char *transport_type,
- const char *local_name, int local_port);
+int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name,
+ int src_port, char *transport_type, const char *local_name, int local_port);
/*!
* \brief General purpose method for creating a SIP request
@@ -2752,4 +2780,54 @@ void ast_sip_modify_id_header(pj_pool_t *pool, pjsip_fromto_hdr *id_hdr,
void ast_sip_get_unidentified_request_thresholds(unsigned int *count, unsigned int *period,
unsigned int *prune_interval);
+/*!
+ * \brief Get the transport name from an endpoint or request uri
+ * \since 13.15.0
+ *
+ * \param endpoint
+ * \param sip_uri
+ * \param buf Buffer to receive transport name
+ * \param buf_len Buffer length
+ *
+ * \retval 0 Success
+ * \retval -1 Failure
+ *
+ * \note
+ * If endpoint->transport is not NULL, it is returned in buf.
+ * Otherwise if sip_uri has an 'x-ast-txp' parameter AND the sip_uri host is
+ * an ip4 or ip6 address, its value is returned,
+ */
+int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, char *buf, size_t buf_len);
+
+/*!
+ * \brief Sets pjsip_tpselector from an endpoint or uri
+ * \since 13.15.0
+ *
+ * \param endpoint If endpoint->transport is set, it's used
+ * \param sip_uri If sip_uri contains a x-ast-txp parameter, it's used
+ * \param selector The selector to be populated
+ *
+ * \retval 0 success
+ * \retval -1 failure
+ */
+int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, pjsip_tpselector *selector);
+
+/*!
+ * \brief Set the transport on a dialog
+ * \since 13.15.0
+ *
+ * \param endpoint
+ * \param dlg
+ * \param selector (optional)
+ *
+ * \note
+ * This API calls ast_sip_get_transport_name(endpoint, dlg->target) and if the result is
+ * non-NULL, calls pjsip_dlg_set_transport. If 'selector' is non-NULL, it is updated with
+ * the selector used.
+ */
+int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
+ pjsip_tpselector *selector);
+
#endif /* _RES_PJSIP_H */
diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 7e65e6d7c..c41cc3ab9 100644
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -83,6 +83,8 @@ struct ast_sip_session_media {
int timeout_sched_id;
/*! \brief Stream is on hold */
unsigned int held:1;
+ /*! \brief Does remote support rtcp_mux */
+ unsigned int remote_rtcp_mux:1;
/*! \brief Stream type this session media handles */
char stream_type[1];
};
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index c4a5b6b41..e8f3d78b4 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -237,6 +237,15 @@ enum ast_rtp_instance_stat {
AST_RTP_INSTANCE_STAT_RXOCTETCOUNT,
};
+enum ast_rtp_instance_rtcp {
+ /*! RTCP should not be sent/received */
+ AST_RTP_INSTANCE_RTCP_DISABLED = 0,
+ /*! RTCP should be sent/received based on standard port rules */
+ AST_RTP_INSTANCE_RTCP_STANDARD,
+ /*! RTCP should be sent/received on the same port as RTP */
+ AST_RTP_INSTANCE_RTCP_MUX,
+};
+
/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
/*! DTMF (RFC2833) */
#define AST_RTP_DTMF (1 << 0)
@@ -443,6 +452,8 @@ struct ast_rtp_engine_ice {
void (*turn_request)(struct ast_rtp_instance *instance, enum ast_rtp_ice_component_type component,
enum ast_transport transport, const char *server, unsigned int port,
const char *username, const char *password);
+ /*! Callback to alter the number of ICE components on a session */
+ void (*change_components)(struct ast_rtp_instance *instance, int num_components);
};
/*! \brief DTLS setup types */
diff --git a/main/http.c b/main/http.c
index 155b04b78..80c7b3cb4 100644
--- a/main/http.c
+++ b/main/http.c
@@ -1915,8 +1915,7 @@ static int httpd_process_request(struct ast_tcptls_session_instance *ser)
static void *httpd_helper_thread(void *data)
{
struct ast_tcptls_session_instance *ser = data;
- struct protoent *p;
- int flags;
+ int flags = 1;
int timeout;
if (!ser || !ser->f) {
@@ -1936,17 +1935,8 @@ static void *httpd_helper_thread(void *data)
* This is necessary to prevent delays (caused by buffering) as we
* write to the socket in bits and pieces.
*/
- p = getprotobyname("tcp");
- if (p) {
- int arg = 1;
-
- if (setsockopt(ser->fd, p->p_proto, TCP_NODELAY, (char *) &arg, sizeof(arg) ) < 0) {
- ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on HTTP connection: %s\n", strerror(errno));
- ast_log(LOG_WARNING, "Some HTTP requests may be slow to respond.\n");
- }
- } else {
- ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on HTTP connection, getprotobyname(\"tcp\") failed\n");
- ast_log(LOG_WARNING, "Some HTTP requests may be slow to respond.\n");
+ if (setsockopt(ser->fd, IPPROTO_TCP, TCP_NODELAY, (char *) &flags, sizeof(flags)) < 0) {
+ ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on HTTP connection: %s\n", strerror(errno));
}
/* make sure socket is non-blocking */
diff --git a/main/manager.c b/main/manager.c
index b8dbb1a04..0f7adf0c8 100644
--- a/main/manager.c
+++ b/main/manager.c
@@ -6591,10 +6591,9 @@ static void *session_do(void *data)
struct mansession s = {
.tcptls_session = data,
};
- int flags;
+ int flags = 1;
int res;
struct ast_sockaddr ser_remote_address_tmp;
- struct protoent *p;
if (ast_atomic_fetchadd_int(&unauth_sessions, +1) >= authlimit) {
fclose(ser->f);
@@ -6614,14 +6613,8 @@ static void *session_do(void *data)
/* here we set TCP_NODELAY on the socket to disable Nagle's algorithm.
* This is necessary to prevent delays (caused by buffering) as we
* write to the socket in bits and pieces. */
- p = getprotobyname("tcp");
- if (p) {
- int arg = 1;
- if( setsockopt(ser->fd, p->p_proto, TCP_NODELAY, (char *)&arg, sizeof(arg) ) < 0 ) {
- ast_log(LOG_WARNING, "Failed to set manager tcp connection to TCP_NODELAY mode: %s\nSome manager actions may be slow to respond.\n", strerror(errno));
- }
- } else {
- ast_log(LOG_WARNING, "Failed to set manager tcp connection to TCP_NODELAY, getprotobyname(\"tcp\") failed\nSome manager actions may be slow to respond.\n");
+ if (setsockopt(ser->fd, IPPROTO_TCP, TCP_NODELAY, (char *) &flags, sizeof(flags)) < 0) {
+ ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on manager connection: %s\n", strerror(errno));
}
/* make sure socket is non-blocking */
diff --git a/main/pbx.c b/main/pbx.c
index d648084d7..df99940ba 100644
--- a/main/pbx.c
+++ b/main/pbx.c
@@ -616,7 +616,7 @@ static int ast_add_extension2_lockopt(struct ast_context *con,
static struct ast_context *find_context_locked(const char *context);
static struct ast_context *find_context(const char *context);
static void get_device_state_causing_channels(struct ao2_container *c);
-static int ext_strncpy(char *dst, const char *src, int len, int nofluff);
+static unsigned int ext_strncpy(char *dst, const char *src, size_t dst_size, int nofluff);
/*!
* \internal
@@ -6907,32 +6907,51 @@ int ast_async_goto_by_name(const char *channame, const char *context, const char
return res;
}
-/*! \brief copy a string skipping whitespace and dashes */
-static int ext_strncpy(char *dst, const char *src, int len, int nofluff)
+/*!
+ * \internal
+ * \brief Copy a string skipping whitespace and optionally dashes.
+ *
+ * \param dst Destination buffer to copy src string.
+ * \param src Null terminated string to copy.
+ * \param dst_size Number of bytes in the dst buffer.
+ * \param nofluf Nonzero if '-' chars are not copied.
+ *
+ * \return Number of bytes written to dst including null terminator.
+ */
+static unsigned int ext_strncpy(char *dst, const char *src, size_t dst_size, int nofluff)
{
- int count = 0;
- int insquares = 0;
+ unsigned int count;
+ unsigned int insquares;
+ unsigned int is_pattern;
- while (*src && (count < len - 1)) {
+ if (!dst_size--) {
+ /* There really is no dst buffer */
+ return 0;
+ }
+
+ count = 0;
+ insquares = 0;
+ is_pattern = *src == '_';
+ while (*src && count < dst_size) {
if (*src == '[') {
- insquares = 1;
+ if (is_pattern) {
+ insquares = 1;
+ }
} else if (*src == ']') {
insquares = 0;
} else if (*src == ' ' && !insquares) {
- src++;
+ ++src;
continue;
} else if (*src == '-' && !insquares && nofluff) {
- src++;
+ ++src;
continue;
}
- *dst = *src;
- dst++;
- src++;
- count++;
+ *dst++ = *src++;
+ ++count;
}
*dst = '\0';
- return count;
+ return count + 1;
}
/*!
@@ -7246,10 +7265,10 @@ static int ast_add_extension2_lockopt(struct ast_context *con,
p += strlen(label) + 1;
}
tmp->name = p;
- p += ext_strncpy(p, extension, strlen(extension) + 1, 0) + 1;
+ p += ext_strncpy(p, extension, strlen(extension) + 1, 0);
if (exten_fluff) {
tmp->exten = p;
- p += ext_strncpy(p, extension, strlen(extension) + 1, 1) + 1;
+ p += ext_strncpy(p, extension, strlen(extension) + 1 - exten_fluff, 1);
} else {
/* no fluff, we don't need a copy. */
tmp->exten = tmp->name;
@@ -7259,10 +7278,10 @@ static int ast_add_extension2_lockopt(struct ast_context *con,
/* Blank callerid and NULL callerid are two SEPARATE things. Do NOT confuse the two!!! */
if (callerid) {
- p += ext_strncpy(p, callerid, strlen(callerid) + 1, 0) + 1;
+ p += ext_strncpy(p, callerid, strlen(callerid) + 1, 0);
if (callerid_fluff) {
tmp->cidmatch = p;
- p += ext_strncpy(p, callerid, strlen(callerid) + 1, 1) + 1;
+ p += ext_strncpy(p, callerid, strlen(callerid) + 1 - callerid_fluff, 1);
}
tmp->matchcid = AST_EXT_MATCHCID_ON;
} else {
diff --git a/res/res_hep.c b/res/res_hep.c
index 15e779012..8d4987c03 100644
--- a/res/res_hep.c
+++ b/res/res_hep.c
@@ -441,6 +441,9 @@ struct hepv3_capture_info *hepv3_create_capture_info(const void *payload, size_t
memcpy(info->payload, payload, len);
info->len = len;
+ /* Set a reasonable default */
+ info->protocol_id = IPPROTO_UDP;
+
return info;
}
@@ -472,7 +475,7 @@ static int hep_queue_cb(void *data)
/* Build HEPv3 header, capture info, and calculate the total packet size */
memcpy(hg_pkt.header.id, "\x48\x45\x50\x33", 4);
- INITIALIZE_GENERIC_HEP_CHUNK_DATA(&hg_pkt.ip_proto, CHUNK_TYPE_IP_PROTOCOL_ID, 0x11);
+ INITIALIZE_GENERIC_HEP_CHUNK_DATA(&hg_pkt.ip_proto, CHUNK_TYPE_IP_PROTOCOL_ID, capture_info->protocol_id);
INITIALIZE_GENERIC_HEP_CHUNK_DATA(&hg_pkt.src_port, CHUNK_TYPE_SRC_PORT, htons(ast_sockaddr_port(&capture_info->src_addr)));
INITIALIZE_GENERIC_HEP_CHUNK_DATA(&hg_pkt.dst_port, CHUNK_TYPE_DST_PORT, htons(ast_sockaddr_port(&capture_info->dst_addr)));
INITIALIZE_GENERIC_HEP_CHUNK_DATA(&hg_pkt.time_sec, CHUNK_TYPE_TIMESTAMP_SEC, htonl(capture_info->capture_time.tv_sec));
diff --git a/res/res_hep_pjsip.c b/res/res_hep_pjsip.c
index 8f5baa2cb..1614b4319 100644
--- a/res/res_hep_pjsip.c
+++ b/res/res_hep_pjsip.c
@@ -73,6 +73,15 @@ static char *assign_uuid(const pj_str_t *call_id, const pj_str_t *local_tag, con
return uuid;
}
+static int transport_to_protocol_id(pjsip_transport *tp)
+{
+ /* XXX If we ever add SCTP support, we'll need to revisit */
+ if (tp->flag & PJSIP_TRANSPORT_RELIABLE) {
+ return IPPROTO_TCP;
+ }
+ return IPPROTO_UDP;
+}
+
static pj_status_t logging_on_tx_msg(pjsip_tx_data *tdata)
{
char local_buf[256];
@@ -126,6 +135,7 @@ static pj_status_t logging_on_tx_msg(pjsip_tx_data *tdata)
ast_sockaddr_parse(&capture_info->src_addr, local_buf, PARSE_PORT_REQUIRE);
ast_sockaddr_parse(&capture_info->dst_addr, remote_buf, PARSE_PORT_REQUIRE);
+ capture_info->protocol_id = transport_to_protocol_id(tdata->tp_info.transport);
capture_info->capture_time = ast_tvnow();
capture_info->capture_type = HEPV3_CAPTURE_TYPE_SIP;
capture_info->uuid = uuid;
@@ -185,6 +195,8 @@ static pj_bool_t logging_on_rx_msg(pjsip_rx_data *rdata)
ast_sockaddr_parse(&capture_info->src_addr, remote_buf, PARSE_PORT_REQUIRE);
ast_sockaddr_parse(&capture_info->dst_addr, local_buf, PARSE_PORT_REQUIRE);
+
+ capture_info->protocol_id = transport_to_protocol_id(rdata->tp_info.transport);
capture_info->capture_time.tv_sec = rdata->pkt_info.timestamp.sec;
capture_info->capture_time.tv_usec = rdata->pkt_info.timestamp.msec * 1000;
capture_info->capture_type = HEPV3_CAPTURE_TYPE_SIP;
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index 347658f9c..7b10f47f6 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -944,6 +944,16 @@
to the receiving one.
</para></description>
</configOption>
+ <configOption name="rtcp_mux" default="no">
+ <synopsis>Enable RFC 5761 RTCP multiplexing on the RTP port</synopsis>
+ <description><para>
+ With this option enabled, Asterisk will attempt to negotiate the use of the "rtcp-mux"
+ attribute on all media streams. This will result in RTP and RTCP being sent and received
+ on the same port. This shifts the demultiplexing logic to the application rather than
+ the transport layer. This option is useful when interoperating with WebRTC endpoints
+ since they mandate this option's use.
+ </para></description>
+ </configOption>
</configObject>
<configObject name="auth">
<synopsis>Authentication type</synopsis>
@@ -1177,6 +1187,22 @@
in-progress calls.</para>
</description>
</configOption>
+ <configOption name="symmetric_transport" default="no">
+ <synopsis>Use the same transport for outgoing reqests as incoming ones.</synopsis>
+ <description>
+ <para>When a request from a dynamic contact
+ comes in on a transport with this option set to 'yes',
+ the transport name will be saved and used for subsequent
+ outgoing requests like OPTIONS, NOTIFY and INVITE. It's
+ saved as a contact uri parameter named 'x-ast-txp' and will
+ display with the contact uri in CLI, AMI, and ARI output.
+ On the outgoing request, if a transport wasn't explicitly
+ set on the endpoint AND the request URI is not a hostname,
+ the saved transport will be used and the 'x-ast-txp'
+ parameter stripped from the outgoing packet.
+ </para>
+ </description>
+ </configOption>
</configObject>
<configObject name="contact">
<synopsis>A way of creating an aliased name to a SIP URI</synopsis>
@@ -2750,7 +2776,54 @@ pjsip_endpoint *ast_sip_get_pjsip_endpoint(void)
return ast_pjsip_endpoint;
}
-static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user, const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
+int ast_sip_get_transport_name(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, char *buf, size_t buf_len)
+{
+ char *host = NULL;
+ static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
+ pjsip_param *x_transport;
+
+ if (!ast_strlen_zero(endpoint->transport)) {
+ ast_copy_string(buf, endpoint->transport, buf_len);
+ return 0;
+ }
+
+ x_transport = pjsip_param_find(&sip_uri->other_param, &x_name);
+ if (!x_transport) {
+ return -1;
+ }
+
+ /* Only use x_transport if the uri host is an ip (4 or 6) address */
+ host = ast_alloca(sip_uri->host.slen + 1);
+ ast_copy_pj_str(host, &sip_uri->host, sip_uri->host.slen + 1);
+ if (!ast_sockaddr_parse(NULL, host, PARSE_PORT_FORBID)) {
+ return -1;
+ }
+
+ ast_copy_pj_str(buf, &x_transport->value, buf_len);
+
+ return 0;
+}
+
+int ast_sip_dlg_set_transport(const struct ast_sip_endpoint *endpoint, pjsip_dialog *dlg,
+ pjsip_tpselector *selector)
+{
+ pjsip_sip_uri *uri;
+ pjsip_tpselector sel = { .type = PJSIP_TPSELECTOR_NONE, };
+
+ uri = pjsip_uri_get_uri(dlg->target);
+ if (!selector) {
+ selector = &sel;
+ }
+
+ ast_sip_set_tpselector_from_ep_or_uri(endpoint, uri, selector);
+ pjsip_dlg_set_transport(dlg, selector);
+
+ return 0;
+}
+
+static int sip_dialog_create_from(pj_pool_t *pool, pj_str_t *from, const char *user,
+ const char *domain, const pj_str_t *target, pjsip_tpselector *selector)
{
pj_str_t tmp, local_addr;
pjsip_uri *uri;
@@ -2880,15 +2953,16 @@ int ast_sip_set_tpselector_from_transport_name(const char *transport_name, pjsip
return ast_sip_set_tpselector_from_transport(transport, selector);
}
-static int sip_get_tpselector_from_endpoint(const struct ast_sip_endpoint *endpoint, pjsip_tpselector *selector)
+int ast_sip_set_tpselector_from_ep_or_uri(const struct ast_sip_endpoint *endpoint,
+ pjsip_sip_uri *sip_uri, pjsip_tpselector *selector)
{
- const char *transport_name = endpoint->transport;
+ char transport_name[128];
- if (ast_strlen_zero(transport_name)) {
+ if (ast_sip_get_transport_name(endpoint, sip_uri, transport_name, sizeof(transport_name))) {
return 0;
}
- return ast_sip_set_tpselector_from_transport_name(endpoint->transport, selector);
+ return ast_sip_set_tpselector_from_transport_name(transport_name, selector);
}
void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t *pool, pjsip_uri *uri)
@@ -2896,8 +2970,8 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
pjsip_sip_uri *sip_uri;
int i = 0;
pjsip_param *param;
- const pj_str_t STR_USER = { "user", 4 };
- const pj_str_t STR_PHONE = { "phone", 5 };
+ static const pj_str_t STR_USER = { "user", 4 };
+ static const pj_str_t STR_PHONE = { "phone", 5 };
if (!endpoint || !endpoint->usereqphone || (!PJSIP_URI_SCHEME_IS_SIP(uri) && !PJSIP_URI_SCHEME_IS_SIPS(uri))) {
return;
@@ -2930,7 +3004,8 @@ void ast_sip_add_usereqphone(const struct ast_sip_endpoint *endpoint, pj_pool_t
pj_list_insert_before(&sip_uri->other_param, param);
}
-pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint, const char *uri, const char *request_user)
+pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
+ const char *uri, const char *request_user)
{
char enclosed_uri[PJSIP_MAX_URL_SIZE];
pj_str_t local_uri = { "sip:temp@temp", 13 }, remote_uri, target_uri;
@@ -2955,12 +3030,13 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
return NULL;
}
- if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
- pjsip_dlg_terminate(dlg);
- return NULL;
- }
+ /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
+ dlg->sess_count++;
+
+ ast_sip_dlg_set_transport(endpoint, dlg, &selector);
if (sip_dialog_create_from(dlg->pool, &local_uri, endpoint->fromuser, endpoint->fromdomain, &remote_uri, &selector)) {
+ dlg->sess_count--;
pjsip_dlg_terminate(dlg);
return NULL;
}
@@ -2996,11 +3072,6 @@ pjsip_dialog *ast_sip_create_dialog_uac(const struct ast_sip_endpoint *endpoint,
ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->target);
ast_sip_add_usereqphone(endpoint, dlg->pool, dlg->remote.info->uri);
- /* We have to temporarily bump up the sess_count here so the dialog is not prematurely destroyed */
- dlg->sess_count++;
-
- pjsip_dlg_set_transport(dlg, &selector);
-
if (!ast_strlen_zero(outbound_proxy)) {
pjsip_route_hdr route_set, *route;
static const pj_str_t ROUTE_HNAME = { "Route", 5 };
@@ -3069,10 +3140,13 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
pjsip_transport_type_e type = rdata->tp_info.transport->key.type;
pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
pjsip_transport *transport;
+ pjsip_contact_hdr *contact_hdr;
ast_assert(status != NULL);
- if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
+ contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+ if (ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(contact_hdr->uri),
+ &selector)) {
return NULL;
}
@@ -3118,8 +3192,8 @@ pjsip_dialog *ast_sip_create_dialog_uas(const struct ast_sip_endpoint *endpoint,
return dlg;
}
-int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
- char *transport_type, const char *local_name, int local_port)
+int ast_sip_create_rdata_with_contact(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
+ char *transport_type, const char *local_name, int local_port, const char *contact)
{
pj_str_t tmp;
@@ -3143,6 +3217,16 @@ int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_nam
return -1;
}
+ if (!ast_strlen_zero(contact)) {
+ pjsip_contact_hdr *contact_hdr;
+
+ contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+ if (contact_hdr) {
+ contact_hdr->uri = pjsip_parse_uri(rdata->tp_info.pool, (char *)contact,
+ strlen(contact), PJSIP_PARSE_URI_AS_NAMEADDR);
+ }
+ }
+
pj_strdup2(rdata->tp_info.pool, &rdata->msg_info.via->recvd_param, rdata->pkt_info.src_name);
rdata->msg_info.via->rport_param = -1;
@@ -3154,6 +3238,13 @@ int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_nam
return 0;
}
+int ast_sip_create_rdata(pjsip_rx_data *rdata, char *packet, const char *src_name, int src_port,
+ char *transport_type, const char *local_name, int local_port)
+{
+ return ast_sip_create_rdata_with_contact(rdata, packet, src_name, src_port, transport_type,
+ local_name, local_port, NULL);
+}
+
/* PJSIP doesn't know about the INFO method, so we have to define it ourselves */
static const pjsip_method info_method = {PJSIP_OTHER_METHOD, {"INFO", 4} };
static const pjsip_method message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
@@ -3235,14 +3326,6 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
pj_cstr(&remote_uri, uri);
}
- if (endpoint) {
- if (sip_get_tpselector_from_endpoint(endpoint, &selector)) {
- ast_log(LOG_ERROR, "Unable to retrieve PJSIP transport selector for endpoint %s\n",
- ast_sorcery_object_get_id(endpoint));
- return -1;
- }
- }
-
pool = pjsip_endpt_create_pool(ast_sip_get_pjsip_endpoint(), "Outbound request", 256, 256);
if (!pool) {
@@ -3260,6 +3343,8 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
return -1;
}
+ ast_sip_set_tpselector_from_ep_or_uri(endpoint, pjsip_uri_get_uri(sip_uri), &selector);
+
fromuser = endpoint ? (!ast_strlen_zero(endpoint->fromuser) ? endpoint->fromuser : ast_sorcery_object_get_id(endpoint)) : NULL;
if (sip_dialog_create_from(pool, &from, fromuser,
endpoint ? endpoint->fromdomain : NULL, &remote_uri, &selector)) {
@@ -3279,6 +3364,8 @@ static int create_out_of_dialog_request(const pjsip_method *method, struct ast_s
return -1;
}
+ pjsip_tx_data_set_transport(*tdata, &selector);
+
if (endpoint && !ast_strlen_zero(endpoint->contact_user)){
pjsip_contact_hdr *contact_hdr;
pjsip_sip_uri *contact_uri;
@@ -3320,6 +3407,8 @@ int ast_sip_create_request(const char *method, struct pjsip_dialog *dlg,
{
const pjsip_method *pmethod = get_pjsip_method(method);
+ ast_assert(endpoint != NULL);
+
if (!pmethod) {
ast_log(LOG_WARNING, "Unknown method '%s'. Cannot send request\n", method);
return -1;
@@ -3584,7 +3673,6 @@ static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
struct send_request_wrapper *req_wrapper;
pj_status_t ret_val;
pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
- pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
if (!cb && token) {
/* Silly. Without a callback we cannot do anything with token. */
@@ -3609,11 +3697,6 @@ static pj_status_t endpt_send_request(struct ast_sip_endpoint *endpoint,
/* Add a reference to tdata. The wrapper destructor cleans it up. */
pjsip_tx_data_add_ref(tdata);
- if (endpoint) {
- sip_get_tpselector_from_endpoint(endpoint, &selector);
- pjsip_tx_data_set_transport(tdata, &selector);
- }
-
if (timeout > 0) {
pj_time_val timeout_timer_val = { timeout / 1000, timeout % 1000 };
diff --git a/res/res_pjsip/config_transport.c b/res/res_pjsip/config_transport.c
index 60b4507cd..3c41f175a 100644
--- a/res/res_pjsip/config_transport.c
+++ b/res/res_pjsip/config_transport.c
@@ -552,13 +552,20 @@ static int transport_apply(const struct ast_sorcery *sorcery, void *obj)
}
}
- if (res == PJ_SUCCESS && (transport->tos || transport->cos)) {
- pj_sock_t sock;
- pj_qos_params qos_params;
- sock = pjsip_udp_transport_get_socket(temp_state->state->transport);
- pj_sock_get_qos_params(sock, &qos_params);
- set_qos(transport, &qos_params);
- pj_sock_set_qos_params(sock, &qos_params);
+ if (res == PJ_SUCCESS) {
+ temp_state->state->transport->info = pj_pool_alloc(temp_state->state->transport->pool,
+ (AST_SIP_X_AST_TXP_LEN + strlen(transport_id) + 2));
+
+ sprintf(temp_state->state->transport->info, "%s:%s", AST_SIP_X_AST_TXP, transport_id);
+
+ if (transport->tos || transport->cos) {
+ pj_sock_t sock;
+ pj_qos_params qos_params;
+ sock = pjsip_udp_transport_get_socket(temp_state->state->transport);
+ pj_sock_get_qos_params(sock, &qos_params);
+ set_qos(transport, &qos_params);
+ pj_sock_set_qos_params(sock, &qos_params);
+ }
}
} else if (transport->type == AST_TRANSPORT_TCP) {
pjsip_tcp_transport_cfg cfg;
@@ -1375,6 +1382,7 @@ int ast_sip_initialize_sorcery_transport(void)
ast_sorcery_object_field_register(sorcery, "transport", "cos", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_transport, cos));
ast_sorcery_object_field_register(sorcery, "transport", "websocket_write_timeout", AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT_STR, OPT_INT_T, PARSE_IN_RANGE, FLDSET(struct ast_sip_transport, write_timeout), 1, INT_MAX);
ast_sorcery_object_field_register(sorcery, "transport", "allow_reload", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_transport, allow_reload));
+ ast_sorcery_object_field_register(sorcery, "transport", "symmetric_transport", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_transport, symmetric_transport));
internal_sip_register_endpoint_formatter(&endpoint_transport_formatter);
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index bfaf750d4..eb8e19712 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -1938,6 +1938,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "subscribe_context", "", OPT_CHAR_ARRAY_T, 0, CHARFLDSET(struct ast_sip_endpoint, subscription.context));
ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "contact_user", "", contact_user_handler, contact_user_to_str, NULL, 0, 0);
ast_sorcery_object_field_register(sip_sorcery, "endpoint", "asymmetric_rtp_codec", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, asymmetric_rtp_codec));
+ ast_sorcery_object_field_register(sip_sorcery, "endpoint", "rtcp_mux", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, rtcp_mux));
if (ast_sip_initialize_sorcery_transport()) {
ast_log(LOG_ERROR, "Failed to register SIP transport support with sorcery\n");
diff --git a/res/res_pjsip/pjsip_message_ip_updater.c b/res/res_pjsip/pjsip_message_ip_updater.c
index 7671ad0a7..864d898b3 100644
--- a/res/res_pjsip/pjsip_message_ip_updater.c
+++ b/res/res_pjsip/pjsip_message_ip_updater.c
@@ -28,6 +28,7 @@
#define MOD_DATA_RESTRICTIONS "restrictions"
static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata);
+static pj_bool_t multihomed_on_rx_message(pjsip_rx_data *rdata);
/*! \brief Outgoing message modification restrictions */
struct multihomed_message_restrictions {
@@ -41,6 +42,7 @@ static pjsip_module multihomed_module = {
.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 1,
.on_tx_request = multihomed_on_tx_message,
.on_tx_response = multihomed_on_tx_message,
+ .on_rx_request = multihomed_on_rx_message,
};
/*! \brief Helper function to get (or allocate if not already present) restrictions on a message */
@@ -151,6 +153,44 @@ static int multihomed_rewrite_sdp(struct pjmedia_sdp_session *sdp)
return 0;
}
+static void sanitize_tdata(pjsip_tx_data *tdata)
+{
+ static const pj_str_t x_name = { AST_SIP_X_AST_TXP, AST_SIP_X_AST_TXP_LEN };
+ pjsip_param *x_transport;
+ pjsip_sip_uri *uri;
+ pjsip_fromto_hdr *fromto;
+ pjsip_contact_hdr *contact;
+ pjsip_hdr *hdr;
+
+ if (tdata->msg->type == PJSIP_REQUEST_MSG) {
+ uri = pjsip_uri_get_uri(tdata->msg->line.req.uri);
+ x_transport = pjsip_param_find(&uri->other_param, &x_name);
+ if (x_transport) {
+ pj_list_erase(x_transport);
+ }
+ }
+
+ for (hdr = tdata->msg->hdr.next; hdr != &tdata->msg->hdr; hdr = hdr->next) {
+ if (hdr->type == PJSIP_H_TO || hdr->type == PJSIP_H_FROM) {
+ fromto = (pjsip_fromto_hdr *) hdr;
+ uri = pjsip_uri_get_uri(fromto->uri);
+ x_transport = pjsip_param_find(&uri->other_param, &x_name);
+ if (x_transport) {
+ pj_list_erase(x_transport);
+ }
+ } else if (hdr->type == PJSIP_H_CONTACT) {
+ contact = (pjsip_contact_hdr *) hdr;
+ uri = pjsip_uri_get_uri(contact->uri);
+ x_transport = pjsip_param_find(&uri->other_param, &x_name);
+ if (x_transport) {
+ pj_list_erase(x_transport);
+ }
+ }
+ }
+
+ pjsip_tx_data_invalidate_msg(tdata);
+}
+
static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
{
struct multihomed_message_restrictions *restrictions = ast_sip_mod_data_get(tdata->mod_data, multihomed_module.id, MOD_DATA_RESTRICTIONS);
@@ -159,6 +199,8 @@ static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
pjsip_via_hdr *via;
pjsip_fromto_hdr *from;
+ sanitize_tdata(tdata);
+
/* Use the destination information to determine what local interface this message will go out on */
pjsip_tpmgr_fla2_param_default(&prm);
prm.tp_type = tdata->tp_info.transport->key.type;
@@ -273,6 +315,47 @@ static pj_status_t multihomed_on_tx_message(pjsip_tx_data *tdata)
return PJ_SUCCESS;
}
+static pj_bool_t multihomed_on_rx_message(pjsip_rx_data *rdata)
+{
+ pjsip_contact_hdr *contact;
+ pjsip_sip_uri *uri;
+ const char *transport_id;
+ struct ast_sip_transport *transport;
+ pjsip_param *x_transport;
+
+ if (rdata->msg_info.msg->type != PJSIP_REQUEST_MSG) {
+ return PJ_FALSE;
+ }
+
+ contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
+ if (!(contact && contact->uri
+ && ast_begins_with(rdata->tp_info.transport->info, AST_SIP_X_AST_TXP ":"))) {
+ return PJ_FALSE;
+ }
+
+ uri = pjsip_uri_get_uri(contact->uri);
+
+ transport_id = rdata->tp_info.transport->info + AST_SIP_X_AST_TXP_LEN + 1;
+ transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_id);
+
+ if (!(transport && transport->symmetric_transport)) {
+ return PJ_FALSE;
+ }
+
+ x_transport = PJ_POOL_ALLOC_T(rdata->tp_info.pool, pjsip_param);
+ x_transport->name = pj_strdup3(rdata->tp_info.pool, AST_SIP_X_AST_TXP);
+ x_transport->value = pj_strdup3(rdata->tp_info.pool, transport_id);
+
+ pj_list_insert_before(&uri->other_param, x_transport);
+
+ ast_debug(1, "Set transport '%s' on %.*s from %.*s:%d\n", transport_id,
+ (int)rdata->msg_info.msg->line.req.method.name.slen,
+ rdata->msg_info.msg->line.req.method.name.ptr,
+ (int)uri->host.slen, uri->host.ptr, uri->port);
+
+ return PJ_FALSE;
+}
+
void ast_res_pjsip_cleanup_message_ip_updater(void)
{
ast_sip_unregister_service(&multihomed_module);
diff --git a/res/res_pjsip_nat.c b/res/res_pjsip_nat.c
index 7404ef5f0..5fcab6378 100644
--- a/res/res_pjsip_nat.c
+++ b/res/res_pjsip_nat.c
@@ -262,32 +262,33 @@ static pj_status_t nat_on_tx_message(pjsip_tx_data *tdata)
return PJ_SUCCESS;
}
- if ( !transport_state->localnet || ast_sockaddr_isnull(&transport_state->external_address)) {
- return PJ_SUCCESS;
- }
-
- ast_sockaddr_parse(&addr, tdata->tp_info.dst_name, PARSE_PORT_FORBID);
- ast_sockaddr_set_port(&addr, tdata->tp_info.dst_port);
+ if (transport_state->localnet) {
+ ast_sockaddr_parse(&addr, tdata->tp_info.dst_name, PARSE_PORT_FORBID);
+ ast_sockaddr_set_port(&addr, tdata->tp_info.dst_port);
- /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */
- if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
- return PJ_SUCCESS;
+ /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */
+ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
+ ast_debug(5, "Request is being sent to local address, skipping NAT manipulation\n");
+ return PJ_SUCCESS;
+ }
}
- /* Update the contact header with the external address */
- if (uri || (uri = nat_get_contact_sip_uri(tdata))) {
- pj_strdup2(tdata->pool, &uri->host, ast_sockaddr_stringify_host(&transport_state->external_address));
- if (transport->external_signaling_port) {
- uri->port = transport->external_signaling_port;
- ast_debug(4, "Re-wrote Contact URI port to %d\n", uri->port);
+ if (!ast_sockaddr_isnull(&transport_state->external_address)) {
+ /* Update the contact header with the external address */
+ if (uri || (uri = nat_get_contact_sip_uri(tdata))) {
+ pj_strdup2(tdata->pool, &uri->host, ast_sockaddr_stringify_host(&transport_state->external_address));
+ if (transport->external_signaling_port) {
+ uri->port = transport->external_signaling_port;
+ ast_debug(4, "Re-wrote Contact URI port to %d\n", uri->port);
+ }
}
- }
- /* Update the via header if relevant */
- if ((tdata->msg->type == PJSIP_REQUEST_MSG) && (via || (via = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_VIA, NULL)))) {
- pj_strdup2(tdata->pool, &via->sent_by.host, ast_sockaddr_stringify_host(&transport_state->external_address));
- if (transport->external_signaling_port) {
- via->sent_by.port = transport->external_signaling_port;
+ /* Update the via header if relevant */
+ if ((tdata->msg->type == PJSIP_REQUEST_MSG) && (via || (via = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_VIA, NULL)))) {
+ pj_strdup2(tdata->pool, &via->sent_by.host, ast_sockaddr_stringify_host(&transport_state->external_address));
+ if (transport->external_signaling_port) {
+ via->sent_by.port = transport->external_signaling_port;
+ }
}
}
diff --git a/res/res_pjsip_pubsub.c b/res/res_pjsip_pubsub.c
index 1892a20e9..79a4a8c3e 100644
--- a/res/res_pjsip_pubsub.c
+++ b/res/res_pjsip_pubsub.c
@@ -123,6 +123,9 @@
<configOption name="expires">
<synopsis>The time at which the subscription expires</synopsis>
</configOption>
+ <configOption name="contact_uri">
+ <synopsis>The Contact URI of the dialog for the subscription</synopsis>
+ </configOption>
</configObject>
<configObject name="resource_list">
<synopsis>Resource list configuration parameters.</synopsis>
@@ -376,6 +379,8 @@ struct subscription_persistence {
char *tag;
/*! When this subscription expires */
struct timeval expires;
+ /*! Contact URI */
+ char contact_uri[PJSIP_MAX_URL_SIZE];
};
/*!
@@ -591,8 +596,8 @@ static void subscription_persistence_update(struct sip_subscription_tree *sub_tr
return;
}
- ast_debug(3, "Updating persistence for '%s->%s'\n",
- ast_sorcery_object_get_id(sub_tree->endpoint), sub_tree->root->resource);
+ ast_debug(3, "Updating persistence for '%s->%s'\n", sub_tree->persistence->endpoint,
+ sub_tree->root->resource);
dlg = sub_tree->dlg;
sub_tree->persistence->cseq = dlg->local.cseq;
@@ -600,10 +605,14 @@ static void subscription_persistence_update(struct sip_subscription_tree *sub_tr
if (rdata) {
int expires;
pjsip_expires_hdr *expires_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_EXPIRES, NULL);
+ pjsip_contact_hdr *contact_hdr = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
expires = expires_hdr ? expires_hdr->ivalue : DEFAULT_PUBLISH_EXPIRES;
sub_tree->persistence->expires = ast_tvadd(ast_tvnow(), ast_samp2tv(expires, 1));
+ pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR, contact_hdr->uri,
+ sub_tree->persistence->contact_uri, sizeof(sub_tree->persistence->contact_uri));
+
/* When receiving a packet on an streaming transport, it's possible to receive more than one SIP
* message at a time into the rdata->pkt_info.packet buffer. However, the rdata->msg_info.msg_buf
* will always point to the proper SIP message that is to be processed. When updating subscription
@@ -1572,8 +1581,9 @@ static int subscription_persistence_recreate(void *obj, void *arg, int flags)
pj_pool_reset(pool);
rdata.tp_info.pool = pool;
- if (ast_sip_create_rdata(&rdata, persistence->packet, persistence->src_name, persistence->src_port,
- persistence->transport_key, persistence->local_name, persistence->local_port)) {
+ if (ast_sip_create_rdata_with_contact(&rdata, persistence->packet, persistence->src_name,
+ persistence->src_port, persistence->transport_key, persistence->local_name,
+ persistence->local_port, persistence->contact_uri)) {
ast_log(LOG_WARNING, "Failed recreating '%s' subscription: The message could not be parsed\n",
persistence->endpoint);
ast_sorcery_delete(ast_sip_get_sorcery(), persistence);
@@ -1725,28 +1735,6 @@ void *ast_sip_subscription_get_header(const struct ast_sip_subscription *sub, co
return pjsip_msg_find_hdr_by_name(msg, &name, NULL);
}
-/*!
- * \internal
- * \brief Wrapper for pjsip_evsub_send_request
- *
- * This function (re)sets the transport before sending to catch cases
- * where the transport might have changed.
- *
- * If pjproject gives us the ability to resend, we'll only reset the transport
- * if PJSIP_ETPNOTAVAIL is returned from send.
- *
- * \returns pj_status_t
- */
-static pj_status_t internal_pjsip_evsub_send_request(struct sip_subscription_tree *sub_tree, pjsip_tx_data *tdata)
-{
- pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
-
- ast_sip_set_tpselector_from_transport_name(sub_tree->endpoint->transport, &selector);
- pjsip_dlg_set_transport(sub_tree->dlg, &selector);
-
- return pjsip_evsub_send_request(sub_tree->evsub, tdata);
-}
-
/* XXX This function is not used. */
struct ast_sip_subscription *ast_sip_create_subscription(const struct ast_sip_subscription_handler *handler,
struct ast_sip_endpoint *endpoint, const char *resource)
@@ -1794,7 +1782,7 @@ struct ast_sip_subscription *ast_sip_create_subscription(const struct ast_sip_su
evsub = sub_tree->evsub;
if (pjsip_evsub_initiate(evsub, NULL, -1, &tdata) == PJ_SUCCESS) {
- internal_pjsip_evsub_send_request(sub_tree, tdata);
+ pjsip_evsub_send_request(sub_tree->evsub, tdata);
} else {
/* pjsip_evsub_terminate will result in pubsub_on_evsub_state,
* being called and terminating the subscription. Therefore, we don't
@@ -1891,7 +1879,7 @@ static int sip_subscription_send_request(struct sip_subscription_tree *sub_tree,
return -1;
}
- res = internal_pjsip_evsub_send_request(sub_tree, tdata);
+ res = pjsip_evsub_send_request(sub_tree->evsub, tdata);
subscription_persistence_update(sub_tree, NULL, SUBSCRIPTION_PERSISTENCE_SEND_REQUEST);
@@ -5343,6 +5331,8 @@ static int load_module(void)
persistence_tag_str2struct, persistence_tag_struct2str, NULL, 0, 0);
ast_sorcery_object_field_register_custom(sorcery, "subscription_persistence", "expires", "",
persistence_expires_str2struct, persistence_expires_struct2str, NULL, 0, 0);
+ ast_sorcery_object_field_register(sorcery, "subscription_persistence", "contact_uri", "", OPT_CHAR_ARRAY_T, 0,
+ CHARFLDSET(struct subscription_persistence, contact_uri));
if (apply_list_configuration(sorcery)) {
ast_sip_unregister_service(&pubsub_module);
diff --git a/res/res_pjsip_refer.c b/res/res_pjsip_refer.c
index d52a922fd..db5061249 100644
--- a/res/res_pjsip_refer.c
+++ b/res/res_pjsip_refer.c
@@ -822,6 +822,13 @@ static int refer_incoming_blind_request(struct ast_sip_session *session, pjsip_r
*/
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
+ /* Uri without exten */
+ if (ast_strlen_zero(exten)) {
+ ast_copy_string(exten, "s", sizeof(exten));
+ ast_debug(3, "Channel '%s' from endpoint '%s' attempted blind transfer to a target without extension. Target was set to 's@%s'\n",
+ ast_channel_name(session->channel), ast_sorcery_object_get_id(session->endpoint), context);
+ }
+
if (!ast_exists_extension(NULL, context, exten, 1, NULL)) {
ast_log(LOG_ERROR, "Channel '%s' from endpoint '%s' attempted blind transfer to '%s@%s' but target does not exist\n",
ast_channel_name(session->channel), ast_sorcery_object_get_id(session->endpoint), exten, context);
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index b27050ed8..d44171cf8 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -169,6 +169,23 @@ static int rtp_check_timeout(const void *data)
return 0;
}
+/*!
+ * \brief Enable RTCP on an RTP session.
+ */
+static void enable_rtcp(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
+ const struct pjmedia_sdp_media *remote_media)
+{
+ enum ast_rtp_instance_rtcp rtcp_type;
+
+ if (session->endpoint->rtcp_mux && session_media->remote_rtcp_mux) {
+ rtcp_type = AST_RTP_INSTANCE_RTCP_MUX;
+ } else {
+ rtcp_type = AST_RTP_INSTANCE_RTCP_STANDARD;
+ }
+
+ ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, rtcp_type);
+}
+
/*! \brief Internal function which creates an RTP instance */
static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
{
@@ -179,6 +196,20 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0);
media_address = &temp_media_address;
+ } else {
+ struct ast_sip_transport *transport =
+ ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
+ session->endpoint->transport);
+
+ if (transport && transport->state) {
+ char hoststr[PJ_INET6_ADDRSTRLEN];
+
+ pj_sockaddr_print(&transport->state->host, hoststr, sizeof(hoststr), 0);
+ ast_debug(1, "Transport: %s bound to host: %s, using this for media.\n",
+ session->endpoint->transport, hoststr);
+ ast_sockaddr_parse(media_address, hoststr, 0);
+ }
+ ao2_cleanup(transport);
}
if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
@@ -186,7 +217,6 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
return -1;
}
- ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, 1);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);
if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
@@ -201,7 +231,7 @@ static int create_rtp(struct ast_sip_session *session, struct ast_sip_session_me
}
if (!strcmp(session_media->stream_type, STR_AUDIO) &&
- (session->endpoint->media.tos_audio || session->endpoint->media.cos_video)) {
+ (session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {
ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
session->endpoint->media.cos_audio, "SIP RTP Audio");
} else if (!strcmp(session_media->stream_type, STR_VIDEO) &&
@@ -555,6 +585,13 @@ static void process_ice_attributes(struct ast_sip_session *session, struct ast_s
continue;
}
+ if (session->endpoint->rtcp_mux && session_media->remote_rtcp_mux && candidate.id > 1) {
+ /* Remote side may have offered RTP and RTCP candidates. However, if we're using RTCP MUX,
+ * then we should ignore RTCP candidates.
+ */
+ continue;
+ }
+
candidate.foundation = foundation;
candidate.transport = transport;
@@ -851,6 +888,26 @@ static int setup_media_encryption(struct ast_sip_session *session,
return 0;
}
+static void set_ice_components(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
+{
+ struct ast_rtp_engine_ice *ice;
+
+ ast_assert(session_media->rtp != NULL);
+
+ ice = ast_rtp_instance_get_ice(session_media->rtp);
+ if (!session->endpoint->media.rtp.ice_support || !ice) {
+ return;
+ }
+
+ if (session->endpoint->rtcp_mux && session_media->remote_rtcp_mux) {
+ /* We both support RTCP mux. Only one ICE component necessary */
+ ice->change_components(session_media->rtp, 1);
+ } else {
+ /* They either don't support RTCP mux or we don't know if they do yet. */
+ ice->change_components(session_media->rtp, 2);
+ }
+}
+
/*! \brief Function which negotiates an incoming media stream */
static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_session *sdp, const struct pjmedia_sdp_media *stream)
@@ -895,6 +952,11 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session, struct
return -1;
}
+ session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(stream, "rtcp-mux", NULL) != NULL);
+ set_ice_components(session, session_media);
+
+ enable_rtcp(session, session_media, stream);
+
res = setup_media_encryption(session, session_media, sdp, stream);
if (res) {
if (!session->endpoint->media.rtp.encryption_optimistic ||
@@ -1065,6 +1127,9 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
return -1;
}
+ set_ice_components(session, session_media);
+ enable_rtcp(session, session_media, NULL);
+
if (!(media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media))) ||
!(media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn)))) {
return -1;
@@ -1228,6 +1293,12 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as
attr->name = STR_SENDRECV;
media->attr[media->attr_count++] = attr;
+ /* If we've got rtcp-mux enabled, just unconditionally offer it in all SDPs */
+ if (session->endpoint->rtcp_mux) {
+ attr = pjmedia_sdp_attr_create(pool, "rtcp-mux", NULL);
+ pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
+ }
+
/* Add the media stream to the SDP */
sdp->media[sdp->media_count++] = media;
@@ -1262,6 +1333,11 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
return -1;
}
+ session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(remote_stream, "rtcp-mux", NULL) != NULL);
+ set_ice_components(session, session_media);
+
+ enable_rtcp(session, session_media, remote_stream);
+
res = setup_media_encryption(session, session_media, remote, remote_stream);
if (!session->endpoint->media.rtp.encryption_optimistic && res) {
/* If optimistic encryption is disabled and crypto should have been enabled but was not
@@ -1293,7 +1369,9 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
return -1;
}
ast_channel_set_fd(session->channel, fdno, ast_rtp_instance_fd(session_media->rtp, 0));
- ast_channel_set_fd(session->channel, fdno + 1, ast_rtp_instance_fd(session_media->rtp, 1));
+ if (!session->endpoint->rtcp_mux || !session_media->remote_rtcp_mux) {
+ ast_channel_set_fd(session->channel, fdno + 1, ast_rtp_instance_fd(session_media->rtp, 1));
+ }
/* If ICE support is enabled find all the needed attributes */
process_ice_attributes(session, session_media, remote, remote_stream);
@@ -1387,10 +1465,11 @@ static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struc
ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID);
/* Is the address within the SDP inside the same network? */
- if (ast_apply_ha(transport_state->localnet, &addr) == AST_SENSE_ALLOW) {
+ if (transport_state->localnet
+ && ast_apply_ha(transport_state->localnet, &addr) == AST_SENSE_ALLOW) {
return;
}
-
+ ast_debug(5, "Setting media address to %s\n", transport->external_media_address);
pj_strdup2(tdata->pool, &stream->conn->addr, transport->external_media_address);
}
diff --git a/res/res_pjsip_session.c b/res/res_pjsip_session.c
index 3c4f102f8..efdce8aa2 100644
--- a/res/res_pjsip_session.c
+++ b/res/res_pjsip_session.c
@@ -973,32 +973,10 @@ int ast_sip_session_refresh(struct ast_sip_session *session,
return 0;
}
-/*!
- * \internal
- * \brief Wrapper for pjsip_inv_send_msg
- *
- * This function (re)sets the transport before sending to catch cases
- * where the transport might have changed.
- *
- * If pjproject gives us the ability to resend, we'll only reset the transport
- * if PJSIP_ETPNOTAVAIL is returned from send.
- *
- * \returns pj_status_t
- */
-static pj_status_t internal_pjsip_inv_send_msg(pjsip_inv_session *inv, const char *transport_name, pjsip_tx_data *tdata)
-{
- pjsip_tpselector selector = { .type = PJSIP_TPSELECTOR_NONE, };
-
- ast_sip_set_tpselector_from_transport_name(transport_name, &selector);
- pjsip_dlg_set_transport(inv->dlg, &selector);
-
- return pjsip_inv_send_msg(inv, tdata);
-}
-
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
{
handle_outgoing_response(session, tdata);
- internal_pjsip_inv_send_msg(session->inv_session, session->endpoint->transport, tdata);
+ pjsip_inv_send_msg(session->inv_session, tdata);
return;
}
@@ -1229,7 +1207,7 @@ void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip
MOD_DATA_ON_RESPONSE, on_response);
handle_outgoing_request(session, tdata);
- internal_pjsip_inv_send_msg(session->inv_session, session->endpoint->transport, tdata);
+ pjsip_inv_send_msg(session->inv_session, tdata);
return;
}
@@ -2049,7 +2027,7 @@ static pjsip_inv_session *pre_session_setup(pjsip_rx_data *rdata, const struct a
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
}
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
return NULL;
}
return inv_session;
@@ -2218,7 +2196,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
}
}
return;
@@ -2230,7 +2208,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
}
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(inv_session);
@@ -2243,7 +2221,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
} else {
- internal_pjsip_inv_send_msg(inv_session, endpoint->transport, tdata);
+ pjsip_inv_send_msg(inv_session, tdata);
}
#ifdef HAVE_PJSIP_INV_SESSION_REF
pjsip_inv_dec_ref(inv_session);
@@ -3112,7 +3090,10 @@ static void session_outgoing_nat_hook(pjsip_tx_data *tdata, struct ast_sip_trans
ast_copy_pj_str(host, &sdp->conn->addr, sizeof(host));
ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID);
- if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) {
+ if (!transport_state->localnet
+ || (transport_state->localnet
+ && ast_apply_ha(transport_state->localnet, &addr) == AST_SENSE_ALLOW)) {
+ ast_debug(5, "Setting external media address to %s\n", transport->external_media_address);
pj_strdup2(tdata->pool, &sdp->conn->addr, transport->external_media_address);
}
}
diff --git a/res/res_pjsip_t38.c b/res/res_pjsip_t38.c
index 0787f0763..16d50cd27 100644
--- a/res/res_pjsip_t38.c
+++ b/res/res_pjsip_t38.c
@@ -869,10 +869,11 @@ static void change_outgoing_sdp_stream_media_address(pjsip_tx_data *tdata, struc
ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID);
/* Is the address within the SDP inside the same network? */
- if (ast_apply_ha(transport_state->localnet, &addr) == AST_SENSE_ALLOW) {
+ if (transport_state->localnet
+ && ast_apply_ha(transport_state->localnet, &addr) == AST_SENSE_ALLOW) {
return;
}
-
+ ast_debug(5, "Setting media address to %s\n", transport->external_media_address);
pj_strdup2(tdata->pool, &stream->conn->addr, transport->external_media_address);
}
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 346db604c..d681fea02 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -331,6 +331,7 @@ struct ast_rtp {
struct ao2_container *ice_active_remote_candidates; /*!< The remote ICE candidates */
struct ao2_container *ice_proposed_remote_candidates; /*!< Incoming remote ICE candidates for new session */
struct ast_sockaddr ice_original_rtp_addr; /*!< rtp address that ICE started on first session */
+ unsigned int ice_num_components; /*!< The number of ICE components */
#endif
#ifdef HAVE_OPENSSL_SRTP
@@ -419,6 +420,7 @@ struct ast_rtcp {
* own address every time
*/
char *local_addr_str;
+ enum ast_rtp_instance_rtcp type;
};
struct rtp_red {
@@ -660,6 +662,22 @@ static int ice_reset_session(struct ast_rtp_instance *instance)
pj_ice_sess_change_role(rtp->ice, role);
}
+ /* If we only have one component now, and we previously set up TURN for RTCP,
+ * we need to destroy that TURN socket.
+ */
+ if (rtp->ice_num_components == 1 && rtp->turn_rtcp) {
+ struct timeval wait = ast_tvadd(ast_tvnow(), ast_samp2tv(TURN_STATE_WAIT_TIME, 1000));
+ struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
+
+ ast_mutex_lock(&rtp->lock);
+ pj_turn_sock_destroy(rtp->turn_rtcp);
+ rtp->turn_state = PJ_TURN_STATE_NULL;
+ while (rtp->turn_state != PJ_TURN_STATE_DESTROYING) {
+ ast_cond_timedwait(&rtp->cond, &rtp->lock, &ts);
+ }
+ ast_mutex_unlock(&rtp->lock);
+ }
+
return res;
}
@@ -775,11 +793,12 @@ static void ast_rtp_ice_start(struct ast_rtp_instance *instance)
ast_log(LOG_WARNING, "No RTP candidates; skipping ICE checklist (%p)\n", instance);
}
- if (!has_rtcp) {
+ /* If we're only dealing with one ICE component, then we don't care about the lack of RTCP candidates */
+ if (!has_rtcp && rtp->ice_num_components > 1) {
ast_log(LOG_WARNING, "No RTCP candidates; skipping ICE checklist (%p)\n", instance);
}
- if (has_rtp && has_rtcp) {
+ if (has_rtp && (has_rtcp || rtp->ice_num_components == 1)) {
pj_status_t res = pj_ice_sess_create_check_list(rtp->ice, &ufrag, &passwd, cand_cnt, &candidates[0]);
char reason[80];
@@ -1271,6 +1290,21 @@ static char *generate_random_string(char *buf, size_t size)
return buf;
}
+static void ast_rtp_ice_change_components(struct ast_rtp_instance *instance, int num_components)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+ /* Don't do anything if ICE is unsupported or if we're not changing the
+ * number of components
+ */
+ if (!icesupport || !rtp->ice || rtp->ice_num_components == num_components) {
+ return;
+ }
+
+ rtp->ice_num_components = num_components;
+ ice_reset_session(instance);
+}
+
/* ICE RTP Engine interface declaration */
static struct ast_rtp_engine_ice ast_rtp_ice = {
.set_authentication = ast_rtp_ice_set_authentication,
@@ -1283,6 +1317,7 @@ static struct ast_rtp_engine_ice ast_rtp_ice = {
.ice_lite = ast_rtp_ice_lite,
.set_role = ast_rtp_ice_set_role,
.turn_request = ast_rtp_ice_turn_request,
+ .change_components = ast_rtp_ice_change_components,
};
#endif
@@ -1542,6 +1577,7 @@ static int ast_rtp_dtls_active(struct ast_rtp_instance *instance)
static void ast_rtp_dtls_stop(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ SSL *ssl = rtp->dtls.ssl;
dtls_srtp_stop_timeout_timer(instance, rtp, 0);
@@ -1559,7 +1595,7 @@ static void ast_rtp_dtls_stop(struct ast_rtp_instance *instance)
if (rtp->rtcp) {
dtls_srtp_stop_timeout_timer(instance, rtp, 1);
- if (rtp->rtcp->dtls.ssl) {
+ if (rtp->rtcp->dtls.ssl && (rtp->rtcp->dtls.ssl != ssl)) {
SSL_free(rtp->rtcp->dtls.ssl);
rtp->rtcp->dtls.ssl = NULL;
ast_mutex_destroy(&rtp->rtcp->dtls.lock);
@@ -1787,7 +1823,7 @@ static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
#ifdef HAVE_OPENSSL_SRTP
dtls_perform_handshake(instance, &rtp->dtls, 0);
- if (rtp->rtcp) {
+ if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
dtls_perform_handshake(instance, &rtp->rtcp->dtls, 1);
}
#endif
@@ -2027,7 +2063,7 @@ static int dtls_srtp_renegotiate(const void *data)
SSL_do_handshake(rtp->dtls.ssl);
dtls_srtp_check_pending(instance, rtp, 0);
- if (rtp->rtcp && rtp->rtcp->dtls.ssl) {
+ if (rtp->rtcp && rtp->rtcp->dtls.ssl && rtp->rtcp->dtls.ssl != rtp->dtls.ssl) {
SSL_renegotiate(rtp->rtcp->dtls.ssl);
SSL_do_handshake(rtp->rtcp->dtls.ssl);
dtls_srtp_check_pending(instance, rtp, 1);
@@ -2618,7 +2654,7 @@ static int ice_create(struct ast_rtp_instance *instance, struct ast_sockaddr *ad
passwd = pj_str(rtp->local_passwd);
/* Create an ICE session for ICE negotiation */
- if (pj_ice_sess_create(&stun_config, NULL, PJ_ICE_SESS_ROLE_UNKNOWN, 2,
+ if (pj_ice_sess_create(&stun_config, NULL, PJ_ICE_SESS_ROLE_UNKNOWN, rtp->ice_num_components,
&ast_rtp_ice_sess_cb, &ufrag, &passwd, NULL, &rtp->ice) == PJ_SUCCESS) {
/* Make this available for the callbacks */
rtp->ice->user_data = instance;
@@ -2627,9 +2663,10 @@ static int ice_create(struct ast_rtp_instance *instance, struct ast_sockaddr *ad
rtp_add_candidates_to_ice(instance, rtp, addr, port, AST_RTP_ICE_COMPONENT_RTP,
TRANSPORT_SOCKET_RTP);
- /* Only add the RTCP candidates to ICE when replacing the session. New sessions
+ /* Only add the RTCP candidates to ICE when replacing the session and if
+ * the ICE session contains more than just an RTP component. New sessions
* handle this in a separate part of the setup phase */
- if (replace && rtp->rtcp) {
+ if (replace && rtp->rtcp && rtp->ice_num_components > 1) {
rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us,
ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP,
TRANSPORT_SOCKET_RTCP);
@@ -2714,6 +2751,7 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
#ifdef HAVE_PJPROJECT
/* Create an ICE session for ICE negotiation */
if (icesupport) {
+ rtp->ice_num_components = 2;
ast_debug(3, "Creating ICE session %s (%d) for RTP instance '%p'\n", ast_sockaddr_stringify(addr), x, instance);
if (ice_create(instance, addr, x, 0)) {
ast_log(LOG_NOTICE, "Failed to start ICE session\n");
@@ -2723,7 +2761,6 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
}
}
#endif
-
/* Record any information we may need */
rtp->sched = sched;
@@ -4154,63 +4191,21 @@ static void update_lost_stats(struct ast_rtp *rtp, unsigned int lost_packets)
rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;
}
-static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
+static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, const unsigned char *rtcpdata, size_t size, struct ast_sockaddr *addr)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
- struct ast_sockaddr addr;
- unsigned char rtcpdata[8192 + AST_FRIENDLY_OFFSET];
- unsigned int *rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
- int res, packetwords, position = 0;
+ unsigned int *rtcpheader = (unsigned int *)(rtcpdata);
+ int packetwords, position = 0;
int report_counter = 0;
struct ast_rtp_rtcp_report_block *report_block;
struct ast_frame *f = &ast_null_frame;
- /* Read in RTCP data from the socket */
- if ((res = rtcp_recvfrom(instance, rtcpdata + AST_FRIENDLY_OFFSET,
- sizeof(rtcpdata) - AST_FRIENDLY_OFFSET,
- 0, &addr)) < 0) {
- ast_assert(errno != EBADF);
- if (errno != EAGAIN) {
- ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n",
- (errno) ? strerror(errno) : "Unspecified");
- return NULL;
- }
- return &ast_null_frame;
- }
-
- /* If this was handled by the ICE session don't do anything further */
- if (!res) {
- return &ast_null_frame;
- }
-
- if (!*(rtcpdata + AST_FRIENDLY_OFFSET)) {
- struct sockaddr_in addr_tmp;
- struct ast_sockaddr addr_v4;
-
- if (ast_sockaddr_is_ipv4(&addr)) {
- ast_sockaddr_to_sin(&addr, &addr_tmp);
- } else if (ast_sockaddr_ipv4_mapped(&addr, &addr_v4)) {
- ast_debug(1, "Using IPv6 mapped address %s for STUN\n",
- ast_sockaddr_stringify(&addr));
- ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
- } else {
- ast_debug(1, "Cannot do STUN for non IPv4 address %s\n",
- ast_sockaddr_stringify(&addr));
- return &ast_null_frame;
- }
- if ((ast_stun_handle_packet(rtp->rtcp->s, &addr_tmp, rtcpdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == AST_STUN_ACCEPT)) {
- ast_sockaddr_from_sin(&addr, &addr_tmp);
- ast_sockaddr_copy(&rtp->rtcp->them, &addr);
- }
- return &ast_null_frame;
- }
-
- packetwords = res / 4;
+ packetwords = size / 4;
if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
/* Send to whoever sent to us */
- if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
- ast_sockaddr_copy(&rtp->rtcp->them, &addr);
+ if (ast_sockaddr_cmp(&rtp->rtcp->them, addr)) {
+ ast_sockaddr_copy(&rtp->rtcp->them, addr);
if (rtpdebug) {
ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
ast_sockaddr_stringify(&rtp->rtcp->them));
@@ -4218,7 +4213,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
}
}
- ast_debug(1, "Got RTCP report of %d bytes\n", res);
+ ast_debug(1, "Got RTCP report of %zu bytes\n", size);
while (position < packetwords) {
int i, pt, rc;
@@ -4246,9 +4241,9 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
return &ast_null_frame;
}
- if (rtcp_debug_test_addr(&addr)) {
+ if (rtcp_debug_test_addr(addr)) {
ast_verbose("\n\nGot RTCP from %s\n",
- ast_sockaddr_stringify(&addr));
+ ast_sockaddr_stringify(addr));
ast_verbose("PT: %d(%s)\n", pt, (pt == RTCP_PT_SR) ? "Sender Report" :
(pt == RTCP_PT_RR) ? "Receiver Report" :
(pt == RTCP_PT_FUR) ? "H.261 FUR" : "Unknown");
@@ -4271,7 +4266,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
(unsigned int)ntohl(rtcpheader[i + 1]),
&rtcp_report->sender_information.ntp_timestamp);
rtcp_report->sender_information.rtp_timestamp = ntohl(rtcpheader[i + 2]);
- if (rtcp_debug_test_addr(&addr)) {
+ if (rtcp_debug_test_addr(addr)) {
ast_verbose("NTP timestamp: %u.%06u\n",
(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
@@ -4303,7 +4298,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
report_block->dlsr = ntohl(rtcpheader[i + 5]);
if (report_block->lsr
&& update_rtt_stats(rtp, report_block->lsr, report_block->dlsr)
- && rtcp_debug_test_addr(&addr)) {
+ && rtcp_debug_test_addr(addr)) {
struct timeval now;
unsigned int lsr_now, lsw, msw;
gettimeofday(&now, NULL);
@@ -4320,7 +4315,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
update_lost_stats(rtp, report_block->lost_count.packets);
rtp->rtcp->reported_jitter_count++;
- if (rtcp_debug_test_addr(&addr)) {
+ if (rtcp_debug_test_addr(addr)) {
ast_verbose(" Fraction lost: %d\n", report_block->lost_count.fraction);
ast_verbose(" Packets lost so far: %u\n", report_block->lost_count.packets);
ast_verbose(" Highest sequence number: %u\n", report_block->highest_seq_no & 0x0000ffff);
@@ -4348,7 +4343,7 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
case RTCP_PT_FUR:
/* Handle RTCP FIR as FUR */
case RTCP_PT_PSFB:
- if (rtcp_debug_test_addr(&addr)) {
+ if (rtcp_debug_test_addr(addr)) {
ast_verbose("Received an RTCP Fast Update Request\n");
}
rtp->f.frametype = AST_FRAME_CONTROL;
@@ -4360,13 +4355,13 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
f = &rtp->f;
break;
case RTCP_PT_SDES:
- if (rtcp_debug_test_addr(&addr)) {
+ if (rtcp_debug_test_addr(addr)) {
ast_verbose("Received an SDES from %s\n",
ast_sockaddr_stringify(&rtp->rtcp->them));
}
break;
case RTCP_PT_BYE:
- if (rtcp_debug_test_addr(&addr)) {
+ if (rtcp_debug_test_addr(addr)) {
ast_verbose("Received a BYE from %s\n",
ast_sockaddr_stringify(&rtp->rtcp->them));
}
@@ -4381,6 +4376,58 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
rtp->rtcp->rtcp_info = 1;
return f;
+
+}
+
+static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
+{
+ struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+ struct ast_sockaddr addr;
+ unsigned char rtcpdata[8192 + AST_FRIENDLY_OFFSET];
+ unsigned char *read_area = rtcpdata + AST_FRIENDLY_OFFSET;
+ size_t read_area_size = sizeof(rtcpdata) - AST_FRIENDLY_OFFSET;
+ int res;
+
+ /* Read in RTCP data from the socket */
+ if ((res = rtcp_recvfrom(instance, read_area, read_area_size,
+ 0, &addr)) < 0) {
+ ast_assert(errno != EBADF);
+ if (errno != EAGAIN) {
+ ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n",
+ (errno) ? strerror(errno) : "Unspecified");
+ return NULL;
+ }
+ return &ast_null_frame;
+ }
+
+ /* If this was handled by the ICE session don't do anything further */
+ if (!res) {
+ return &ast_null_frame;
+ }
+
+ if (!*read_area) {
+ struct sockaddr_in addr_tmp;
+ struct ast_sockaddr addr_v4;
+
+ if (ast_sockaddr_is_ipv4(&addr)) {
+ ast_sockaddr_to_sin(&addr, &addr_tmp);
+ } else if (ast_sockaddr_ipv4_mapped(&addr, &addr_v4)) {
+ ast_debug(1, "Using IPv6 mapped address %s for STUN\n",
+ ast_sockaddr_stringify(&addr));
+ ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
+ } else {
+ ast_debug(1, "Cannot do STUN for non IPv4 address %s\n",
+ ast_sockaddr_stringify(&addr));
+ return &ast_null_frame;
+ }
+ if ((ast_stun_handle_packet(rtp->rtcp->s, &addr_tmp, read_area, res, NULL, NULL) == AST_STUN_ACCEPT)) {
+ ast_sockaddr_from_sin(&addr, &addr_tmp);
+ ast_sockaddr_copy(&rtp->rtcp->them, &addr);
+ }
+ return &ast_null_frame;
+ }
+
+ return ast_rtcp_interpret(instance, read_area, res, &addr);
}
static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int *rtpheader, int len, int hdrlen)
@@ -4487,19 +4534,54 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int
return 0;
}
+static int rtcp_mux(struct ast_rtp *rtp, const unsigned char *packet)
+{
+ uint8_t version;
+ uint8_t pt;
+ uint8_t m;
+
+ if (!rtp->rtcp || rtp->rtcp->type != AST_RTP_INSTANCE_RTCP_MUX) {
+ return 0;
+ }
+
+ version = (packet[0] & 0XC0) >> 6;
+ if (version == 0) {
+ /* version 0 indicates this is a STUN packet and shouldn't
+ * be interpreted as a possible RTCP packet
+ */
+ return 0;
+ }
+
+ /* The second octet of a packet will be one of the following:
+ * For RTP: The marker bit (1 bit) and the RTP payload type (7 bits)
+ * For RTCP: The payload type (8)
+ *
+ * RTP has a forbidden range of payload types (64-95) since these
+ * will conflict with RTCP payload numbers if the marker bit is set.
+ */
+ m = packet[1] & 0x80;
+ pt = packet[1] & 0x7F;
+ if (m && pt >= 64 && pt <= 95) {
+ return 1;
+ }
+ return 0;
+}
+
static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
struct ast_sockaddr addr;
int res, hdrlen = 12, version, payloadtype, padding, mark, ext, cc, prev_seqno;
- unsigned int *rtpheader = (unsigned int*)(rtp->rawdata + AST_FRIENDLY_OFFSET), seqno, ssrc, timestamp;
+ unsigned char *read_area = rtp->rawdata + AST_FRIENDLY_OFFSET;
+ size_t read_area_size = sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET;
+ unsigned int *rtpheader = (unsigned int*)(read_area), seqno, ssrc, timestamp;
RAII_VAR(struct ast_rtp_payload_type *, payload, NULL, ao2_cleanup);
struct ast_sockaddr remote_address = { {0,} };
struct frame_list frames;
/* If this is actually RTCP let's hop on over and handle it */
if (rtcp) {
- if (rtp->rtcp) {
+ if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
return ast_rtcp_read(instance);
}
return &ast_null_frame;
@@ -4511,8 +4593,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
}
/* Actually read in the data from the socket */
- if ((res = rtp_recvfrom(instance, rtp->rawdata + AST_FRIENDLY_OFFSET,
- sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0,
+ if ((res = rtp_recvfrom(instance, read_area, read_area_size, 0,
&addr)) < 0) {
ast_assert(errno != EBADF);
if (errno != EAGAIN) {
@@ -4528,12 +4609,17 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
return &ast_null_frame;
}
+ /* This could be a multiplexed RTCP packet. If so, be sure to interpret it correctly */
+ if (rtcp_mux(rtp, read_area)) {
+ return ast_rtcp_interpret(instance, read_area, res, &addr);
+ }
+
/* Make sure the data that was read in is actually enough to make up an RTP packet */
if (res < hdrlen) {
/* If this is a keepalive containing only nulls, don't bother with a warning */
int i;
for (i = 0; i < res; ++i) {
- if (rtp->rawdata[AST_FRIENDLY_OFFSET + i] != '\0') {
+ if (read_area[i] != '\0') {
ast_log(LOG_WARNING, "RTP Read too short\n");
return &ast_null_frame;
}
@@ -4560,7 +4646,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
ast_sockaddr_stringify(&addr));
return &ast_null_frame;
}
- if ((ast_stun_handle_packet(rtp->s, &addr_tmp, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == AST_STUN_ACCEPT) &&
+ if ((ast_stun_handle_packet(rtp->s, &addr_tmp, read_area, res, NULL, NULL) == AST_STUN_ACCEPT) &&
ast_sockaddr_isnull(&remote_address)) {
ast_sockaddr_from_sin(&addr, &addr_tmp);
ast_rtp_instance_set_remote_address(instance, &addr);
@@ -4609,7 +4695,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* do not update the originally given address, but only the remote */
ast_rtp_instance_set_incoming_source_address(instance, &addr);
ast_sockaddr_copy(&remote_address, &addr);
- if (rtp->rtcp) {
+ if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
ast_sockaddr_copy(&rtp->rtcp->them, &addr);
ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(&addr) + 1);
}
@@ -4676,7 +4762,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
/* Remove any padding bytes that may be present */
if (padding) {
- res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
+ res -= read_area[res - 1];
}
/* Skip over any CSRC fields */
@@ -4750,11 +4836,11 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
* by passing the pointer to the frame list to it so that the method
* can append frames to the list as needed.
*/
- process_dtmf_rfc2833(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark, &frames);
+ process_dtmf_rfc2833(instance, read_area + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark, &frames);
} else if (payload->rtp_code == AST_RTP_CISCO_DTMF) {
- f = process_dtmf_cisco(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
+ f = process_dtmf_cisco(instance, read_area + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
} else if (payload->rtp_code == AST_RTP_CN) {
- f = process_cn_rfc3389(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
+ f = process_cn_rfc3389(instance, read_area + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n",
payloadtype,
@@ -4810,7 +4896,7 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
rtp->f.src = "RTP";
rtp->f.mallocd = 0;
rtp->f.datalen = res - hdrlen;
- rtp->f.data.ptr = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
+ rtp->f.data.ptr = read_area + hdrlen;
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
rtp->f.seqno = seqno;
@@ -4921,19 +5007,31 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
if (value) {
struct ast_sockaddr local_addr;
- if (rtp->rtcp) {
+ if (rtp->rtcp && rtp->rtcp->type == value) {
ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
return;
}
- /* Setup RTCP to be activated on the next RTP write */
- if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
- return;
+
+ if (!rtp->rtcp) {
+ rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp));
+ if (!rtp->rtcp) {
+ return;
+ }
+ rtp->rtcp->s = -1;
+#ifdef HAVE_OPENSSL_SRTP
+ rtp->rtcp->dtls.timeout_timer = -1;
+#endif
+ rtp->rtcp->schedid = -1;
}
+ rtp->rtcp->type = value;
+
/* Grab the IP address and port we are going to use */
ast_rtp_instance_get_local_address(instance, &rtp->rtcp->us);
- ast_sockaddr_set_port(&rtp->rtcp->us,
- ast_sockaddr_port(&rtp->rtcp->us) + 1);
+ if (value == AST_RTP_INSTANCE_RTCP_STANDARD) {
+ ast_sockaddr_set_port(&rtp->rtcp->us,
+ ast_sockaddr_port(&rtp->rtcp->us) + 1);
+ }
ast_sockaddr_copy(&local_addr, &rtp->rtcp->us);
if (!ast_find_ourip(&local_addr, &rtp->rtcp->us, 0)) {
@@ -4943,6 +5041,7 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
ast_sockaddr_copy(&local_addr, &rtp->rtcp->us);
}
+ ast_free(rtp->rtcp->local_addr_str);
rtp->rtcp->local_addr_str = ast_strdup(ast_sockaddr_stringify(&local_addr));
if (!rtp->rtcp->local_addr_str) {
ast_free(rtp->rtcp);
@@ -4950,43 +5049,67 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
return;
}
- if ((rtp->rtcp->s =
- create_new_socket("RTCP",
- ast_sockaddr_is_ipv4(&rtp->rtcp->us) ?
- AF_INET :
- ast_sockaddr_is_ipv6(&rtp->rtcp->us) ?
- AF_INET6 : -1)) < 0) {
- ast_debug(1, "Failed to create a new socket for RTCP on instance '%p'\n", instance);
- ast_free(rtp->rtcp->local_addr_str);
- ast_free(rtp->rtcp);
- rtp->rtcp = NULL;
- return;
- }
-
- /* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
- if (ast_bind(rtp->rtcp->s, &rtp->rtcp->us)) {
- ast_debug(1, "Failed to setup RTCP on RTP instance '%p'\n", instance);
- close(rtp->rtcp->s);
- ast_free(rtp->rtcp->local_addr_str);
- ast_free(rtp->rtcp);
- rtp->rtcp = NULL;
- return;
- }
-
- ast_debug(1, "Setup RTCP on RTP instance '%p'\n", instance);
- rtp->rtcp->schedid = -1;
+ if (value == AST_RTP_INSTANCE_RTCP_STANDARD) {
+ /* We're either setting up RTCP from scratch or
+ * switching from MUX. Either way, we won't have
+ * a socket set up, and we need to set it up
+ */
+ if ((rtp->rtcp->s =
+ create_new_socket("RTCP",
+ ast_sockaddr_is_ipv4(&rtp->rtcp->us) ?
+ AF_INET :
+ ast_sockaddr_is_ipv6(&rtp->rtcp->us) ?
+ AF_INET6 : -1)) < 0) {
+ ast_debug(1, "Failed to create a new socket for RTCP on instance '%p'\n", instance);
+ ast_free(rtp->rtcp->local_addr_str);
+ ast_free(rtp->rtcp);
+ rtp->rtcp = NULL;
+ return;
+ }
+ /* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
+ if (ast_bind(rtp->rtcp->s, &rtp->rtcp->us)) {
+ ast_debug(1, "Failed to setup RTCP on RTP instance '%p'\n", instance);
+ close(rtp->rtcp->s);
+ ast_free(rtp->rtcp->local_addr_str);
+ ast_free(rtp->rtcp);
+ rtp->rtcp = NULL;
+ return;
+ }
#ifdef HAVE_PJPROJECT
- if (rtp->ice) {
- rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us, ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_SOCKET_RTCP);
- }
+ if (rtp->ice) {
+ rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us, ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_SOCKET_RTCP);
+ }
#endif
-
#ifdef HAVE_OPENSSL_SRTP
- rtp->rtcp->dtls.timeout_timer = -1;
- dtls_setup_rtcp(instance);
+ dtls_setup_rtcp(instance);
#endif
+ } else {
+ struct ast_sockaddr addr;
+ /* RTCPMUX uses the same socket as RTP. If we were previously using standard RTCP
+ * then close the socket we previously created.
+ *
+ * It may seem as though there is a possible race condition here where we might try
+ * to close the RTCP socket while it is being used to send data. However, this is not
+ * a problem in practice since setting and adjusting of RTCP properties happens prior
+ * to activating RTP. It is not until RTP is activated that timers start for RTCP
+ * transmission
+ */
+ if (rtp->rtcp->s > -1) {
+ close(rtp->rtcp->s);
+ }
+ rtp->rtcp->s = rtp->s;
+ ast_rtp_instance_get_remote_address(instance, &addr);
+ ast_sockaddr_copy(&rtp->rtcp->them, &addr);
+#ifdef HAVE_OPENSSL_SRTP
+ if (rtp->rtcp->dtls.ssl) {
+ SSL_free(rtp->rtcp->dtls.ssl);
+ }
+ rtp->rtcp->dtls.ssl = rtp->dtls.ssl;
+#endif
+ }
+ ast_debug(1, "Setup RTCP on RTP instance '%p'\n", instance);
return;
} else {
if (rtp->rtcp) {
@@ -5001,9 +5124,11 @@ static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_pro
}
rtp->rtcp->schedid = -1;
}
- close(rtp->rtcp->s);
+ if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->s) {
+ close(rtp->rtcp->s);
+ }
#ifdef HAVE_OPENSSL_SRTP
- if (rtp->rtcp->dtls.ssl) {
+ if (rtp->rtcp->dtls.ssl && rtp->rtcp->dtls.ssl != rtp->dtls.ssl) {
SSL_free(rtp->rtcp->dtls.ssl);
}
#endif
@@ -5045,10 +5170,12 @@ static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct
ast_debug(1, "Setting RTCP address on RTP instance '%p'\n", instance);
ast_sockaddr_copy(&rtp->rtcp->them, addr);
if (!ast_sockaddr_isnull(addr)) {
- ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(addr) + 1);
+ if (rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
+ ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(addr) + 1);
- /* Update the local RTCP address with what is being used */
- ast_sockaddr_set_port(&local, ast_sockaddr_port(&local) + 1);
+ /* Update the local RTCP address with what is being used */
+ ast_sockaddr_set_port(&local, ast_sockaddr_port(&local) + 1);
+ }
ast_sockaddr_copy(&rtp->rtcp->us, &local);
ast_free(rtp->rtcp->local_addr_str);
@@ -5336,7 +5463,7 @@ static int ast_rtp_activate(struct ast_rtp_instance *instance)
dtls_perform_handshake(instance, &rtp->dtls, 0);
- if (rtp->rtcp) {
+ if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
dtls_perform_handshake(instance, &rtp->rtcp->dtls, 1);
}