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-rw-r--r--channels/chan_sip.c103
1 files changed, 92 insertions, 11 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 74c59822e..940653825 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -543,6 +543,8 @@ static int regobjs = 0; /*!< Registry objects */
static struct ast_flags global_flags[2] = {{0}}; /*!< global SIP_ flags */
+static int global_autoframing = 0;
+
/*! \brief Protect the SIP dialog list (of sip_pvt's) */
AST_MUTEX_DEFINE_STATIC(iflock);
@@ -966,6 +968,7 @@ static struct sip_pvt {
struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
struct sip_pvt *next; /*!< Next dialog in chain */
struct sip_invite_param *options; /*!< Options for INVITE */
+ int autoframing;
} *iflist = NULL;
#define FLAG_RESPONSE (1 << 0)
@@ -1015,6 +1018,7 @@ struct sip_user {
struct ast_ha *ha; /*!< ACL setting */
struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
int maxcallbitrate; /*!< Maximum Bitrate for a video call */
+ int autoframing;
};
/*! \brief Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host) */
@@ -1077,6 +1081,7 @@ struct sip_peer {
struct ast_variable *chanvars; /*!< Variables to set for channel created by user */
struct sip_pvt *mwipvt; /*!< Subscription for MWI */
int lastmsg;
+ int autoframing;
};
@@ -2541,6 +2546,11 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
ast_log(LOG_DEBUG, "Setting NAT on UDPTL to %s\n", natflags ? "On" : "Off");
ast_udptl_setnat(dialog->udptl, natflags);
}
+ /* Set Frame packetization */
+ if (dialog->rtp) {
+ ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
+ dialog->autoframing = peer->autoframing;
+ }
ast_string_field_set(dialog, peername, peer->username);
ast_string_field_set(dialog, authname, peer->username);
ast_string_field_set(dialog, username, peer->username);
@@ -4702,6 +4712,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
*/
/* XXX This needs to be done per media stream, since it's media stream specific */
iterator = req->sdp_start;
+ int found_rtpmap_codecs[32];
+ int last_rtpmap_codec=0;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
if (option_debug > 1) {
@@ -4752,17 +4764,49 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
} else if (!strcasecmp(a, "sendrecv")) {
sendonly = 0;
continue;
- } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2)
+ } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
+ char *tmp = strrchr(a, ':');
+ long int framing = 0;
+ if (tmp) {
+ tmp++;
+ framing = strtol(tmp, NULL, 10);
+ if (framing == LONG_MIN || framing == LONG_MAX) {
+ framing = 0;
+ ast_log(LOG_DEBUG, "Can't read framing from SDP: %s\n", a);
+ }
+ }
+ if (framing && last_rtpmap_codec) {
+ if (p->autoframing || global_autoframing) {
+ struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ int codec_n;
+ int format = 0;
+ for (codec_n = 0; codec_n < last_rtpmap_codec; codec_n++) {
+ format = ast_rtp_codec_getformat(found_rtpmap_codecs[codec_n]);
+ if (!format) /* non-codec or not found */
+ continue;
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
+ ast_codec_pref_setsize(pref, format, framing);
+ }
+ ast_rtp_codec_setpref(p->rtp, pref);
+ }
+ }
+ memset(&found_rtpmap_codecs, 0, sizeof(found_rtpmap_codecs));
+ last_rtpmap_codec = 0;
continue;
- /* We have a rtpmap to handle */
- if (debug)
- ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
+ } else if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
+ /* We have a rtpmap to handle */
+ if (debug)
+ ast_verbose("Found description format %s for ID %d\n", mimeSubtype, codec);
+ found_rtpmap_codecs[last_rtpmap_codec] = codec;
+ last_rtpmap_codec++;
- /* Note: should really look at the 'freq' and '#chans' params too */
- ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
+ /* Note: should really look at the 'freq' and '#chans' params too */
+ ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0);
- if (p->vrtp)
- ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
+ if (p->vrtp)
+ ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0);
+ }
}
if (udptlportno != -1) {
@@ -5593,12 +5637,19 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
int debug)
{
int rtp_code;
+ struct ast_format_list fmt;
+
if (debug)
ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
return;
+ if (p->rtp) {
+ struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+ fmt = ast_codec_pref_getsize(pref, codec);
+ } else /* I dont see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
+ return;
ast_build_string(m_buf, m_size, " %d", rtp_code);
ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype(1, codec,
@@ -5608,9 +5659,12 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
/* Indicate that we don't support VAD (G.729 annex B) */
ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
} else if (codec == AST_FORMAT_ILBC) {
- /* Add information about us using only 20 ms packetization */
- ast_build_string(a_buf, a_size, "a=fmtp:%d mode=20\r\n", rtp_code);
-
+ /* Add information about us using only 20/30 ms packetization */
+ ast_build_string(a_buf, a_size, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
+ }
+
+ if (codec != AST_FORMAT_ILBC) {
+ ast_build_string(a_buf, a_size, "a=ptime:%d\r\n", fmt.cur_ms);
}
}
@@ -6084,6 +6138,11 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
}
respprep(&resp, p, msg, req);
if (p->rtp) {
+ if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Setting framing from config on incoming call\n");
+ ast_rtp_codec_setpref(p->rtp, &p->prefs);
+ }
try_suggested_sip_codec(p);
add_sdp(&resp, p);
} else
@@ -7930,6 +7989,11 @@ static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr
ASTOBJ_UNREF(peer, sip_destroy_peer);
}
if (peer) {
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ }
if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_DYNAMIC)) {
ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
} else {
@@ -8663,6 +8727,11 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
}
}
p->prefs = user->prefs;
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &p->prefs);
+ p->autoframing = user->autoframing;
+ }
/* replace callerid if rpid found, and not restricted */
if (!ast_strlen_zero(rpid_num) && ast_test_flag(&p->flags[0], SIP_TRUSTRPID)) {
char *tmp;
@@ -8771,6 +8840,11 @@ static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_requ
peer = find_peer(NULL, &p->recv, 1);
if (peer) {
+ /* Set Frame packetization */
+ if (p->rtp) {
+ ast_rtp_codec_setpref(p->rtp, &peer->prefs);
+ p->autoframing = peer->autoframing;
+ }
if (debug)
ast_verbose("Found peer '%s'\n", peer->name);
@@ -9528,6 +9602,7 @@ static void print_codec_to_cli(int fd, struct ast_codec_pref *pref)
if (!codec)
break;
ast_cli(fd, "%s", ast_getformatname(codec));
+ ast_cli(fd, ":%d", pref->framing[x]);
if (x < 31 && ast_codec_pref_index(pref, x + 1))
ast_cli(fd, ",");
}
@@ -15167,6 +15242,8 @@ static struct sip_user *build_user(const char *name, struct ast_variable *v, int
ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
ast_parse_allow_disallow(&user->prefs, &user->capability, v->value, 0);
+ } else if (!strcasecmp(v->name, "autoframing")) {
+ user->autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "callingpres")) {
user->callingpres = ast_parse_caller_presentation(v->value);
if (user->callingpres == -1)
@@ -15446,6 +15523,8 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, 0);
+ } else if (!strcasecmp(v->name, "autoframing")) {
+ peer->autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "rtptimeout")) {
if ((sscanf(v->value, "%d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d. Using default.\n", v->value, v->lineno);
@@ -15807,6 +15886,8 @@ static int reload_config(enum channelreloadreason reason)
ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, 0);
+ } else if (!strcasecmp(v->name, "autoframing")) {
+ global_autoframing = ast_true(v->value);
} else if (!strcasecmp(v->name, "allowexternaldomains")) {
allow_external_domains = ast_true(v->value);
} else if (!strcasecmp(v->name, "autodomain")) {
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-10 07:21:41 +0000