diff options
author | Joshua Colp <jcolp@digium.com> | 2014-11-03 14:45:01 +0000 |
---|---|---|
committer | Joshua Colp <jcolp@digium.com> | 2014-11-03 14:45:01 +0000 |
commit | ac091d41844a9a4a0f7d539164bcd154351b6da7 (patch) | |
tree | 84ec4d1350b4e6d1d1498c4ceabd2b5484f3947d /res | |
parent | 285be15aaf0469055d3392ecd73eb24395e49059 (diff) |
chan_pjsip: Add support for passing hold and unhold requests through.
This change adds an option, moh_passthrough, that when enabled will pass
hold and unhold requests through using a SIP re-invite. When placing on
hold a re-invite with sendonly will be sent and when taking off hold a
re-invite with sendrecv will be sent. This allows remote servers to handle
the musiconhold instead of the local Asterisk instance being responsible.
Review: https://reviewboard.asterisk.org/r/4103/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res')
-rw-r--r-- | res/res_pjsip.c | 6 | ||||
-rw-r--r-- | res/res_pjsip/pjsip_configuration.c | 1 | ||||
-rw-r--r-- | res/res_pjsip_sdp_rtp.c | 11 |
3 files changed, 13 insertions, 5 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c index b350b7b77..dcf771bb3 100644 --- a/res/res_pjsip.c +++ b/res/res_pjsip.c @@ -576,6 +576,9 @@ <configOption name="user_eq_phone" default="no"> <synopsis>Determines whether a user=phone parameter is placed into the request URI if the user is determined to be a phone number</synopsis> </configOption> + <configOption name="moh_passthrough" default="no"> + <synopsis>Determines whether hold and unhold will be passed through using re-INVITEs with recvonly and sendrecv to the remote side</synopsis> + </configOption> <configOption name="sdp_owner" default="-"> <synopsis>String placed as the username portion of an SDP origin (o=) line.</synopsis> </configOption> @@ -1560,6 +1563,9 @@ <parameter name="UserEqPhone"> <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='user_eq_phone']/synopsis/node())"/></para> </parameter> + <parameter name="MohPassthrough"> + <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='moh_passthrough']/synopsis/node())"/></para> + </parameter> <parameter name="SdpOwner"> <para><xi:include xpointer="xpointer(/docs/configInfo[@name='res_pjsip']/configFile[@name='pjsip.conf']/configObject[@name='endpoint']/configOption[@name='sdp_owner']/synopsis/node())"/></para> </parameter> diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c index dabbfaed8..798066777 100644 --- a/res/res_pjsip/pjsip_configuration.c +++ b/res/res_pjsip/pjsip_configuration.c @@ -1733,6 +1733,7 @@ int ast_res_pjsip_initialize_configuration(const struct ast_module_info *ast_mod ast_sorcery_object_field_register(sip_sorcery, "endpoint", "record_off_feature", "automixmon", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, info.recording.offfeature)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "allow_transfer", "yes", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, allowtransfer)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "user_eq_phone", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, usereqphone)); + ast_sorcery_object_field_register(sip_sorcery, "endpoint", "moh_passthrough", "no", OPT_BOOL_T, 1, FLDSET(struct ast_sip_endpoint, moh_passthrough)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_owner", "-", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpowner)); ast_sorcery_object_field_register(sip_sorcery, "endpoint", "sdp_session", "Asterisk", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_endpoint, media.sdpsession)); ast_sorcery_object_field_register_custom(sip_sorcery, "endpoint", "tos_audio", "0", tos_handler, tos_audio_to_str, NULL, 0, 0); diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index 1f863008f..74c980d39 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -887,6 +887,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as static const pj_str_t STR_IP4 = { "IP4", 3}; static const pj_str_t STR_IP6 = { "IP6", 3}; static const pj_str_t STR_SENDRECV = { "sendrecv", 8 }; + static const pj_str_t STR_SENDONLY = { "sendonly", 8 }; pjmedia_sdp_media *media; char hostip[PJ_INET6_ADDRSTRLEN+2]; struct ast_sockaddr addr; @@ -1046,7 +1047,7 @@ static int create_outgoing_sdp_stream(struct ast_sip_session *session, struct as /* Add the sendrecv attribute - we purposely don't keep track because pjmedia-sdp will automatically change our offer for us */ attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr); - attr->name = STR_SENDRECV; + attr->name = !session_media->locally_held ? STR_SENDRECV : STR_SENDONLY; media->attr[media->attr_count++] = attr; /* Add the media stream to the SDP */ @@ -1122,18 +1123,18 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a if (ast_sockaddr_isnull(addrs) || ast_sockaddr_is_any(addrs) || pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) { - if (!session_media->held) { + if (!session_media->remotely_held) { /* The remote side has put us on hold */ ast_queue_hold(session->channel, session->endpoint->mohsuggest); ast_rtp_instance_stop(session_media->rtp); ast_queue_frame(session->channel, &ast_null_frame); - session_media->held = 1; + session_media->remotely_held = 1; } - } else if (session_media->held) { + } else if (session_media->remotely_held) { /* The remote side has taken us off hold */ ast_queue_unhold(session->channel); ast_queue_frame(session->channel, &ast_null_frame); - session_media->held = 0; + session_media->remotely_held = 0; } return 1; |