diff options
author | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-30 18:58:49 +0000 |
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committer | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-09-30 18:58:49 +0000 |
commit | 25411e5ee985044deac61738e617d19bf891df61 (patch) | |
tree | 4adcfdf7a2538c95fccfe3522159eb0c75aa0bab /channels | |
parent | cc3947653d564b8cabc8cc6b3bd400421d3ff70e (diff) |
Merged revisions 221266 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
................
r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
........
................
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@221302 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 23 |
1 files changed, 22 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 029c15728..688ff81ef 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1061,6 +1061,7 @@ struct sip_auth { #define SIP_PAGE2_RTCACHEFRIENDS (1 << 0) /*!< GP: Should we keep RT objects in memory for extended time? */ #define SIP_PAGE2_RTAUTOCLEAR (1 << 2) /*!< GP: Should we clean memory from peers after expiry? */ /* Space for addition of other realtime flags in the future */ +#define SIP_PAGE2_CONSTANT_SSRC (1 << 8) /*!< GDP: Don't change SSRC on reinvite */ #define SIP_PAGE2_STATECHANGEQUEUE (1 << 9) /*!< D: Unsent state pending change exists */ #define SIP_PAGE2_RPORT_PRESENT (1 << 10) /*!< Was rport received in the Via header? */ @@ -1092,7 +1093,7 @@ struct sip_auth { (SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \ SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \ SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | \ - SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS) + SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_CONSTANT_SSRC) /*@}*/ @@ -4525,6 +4526,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout); ast_rtp_set_rtpholdtimeout(dialog->rtp, peer->rtpholdtimeout); ast_rtp_set_rtpkeepalive(dialog->rtp, peer->rtpkeepalive); + if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) { + ast_rtp_set_constantssrc(dialog->rtp); + } /* Set Frame packetization */ ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs); dialog->autoframing = peer->autoframing; @@ -4535,6 +4539,9 @@ static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer) ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout); ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout); ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive); + if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_CONSTANT_SSRC)) { + ast_rtp_set_constantssrc(dialog->vrtp); + } } if (dialog->trtp) { /* Realtime text */ ast_rtp_setdtmf(dialog->trtp, 0); @@ -18501,6 +18508,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); return -1; } + ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE); } else { p->jointcapability = p->capability; ast_debug(1, "Hm.... No sdp for the moment\n"); @@ -18549,6 +18557,14 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int ast_debug(1, "No compatible codecs for this SIP call.\n"); return -1; } + if (ast_test_flag(&p->flags[1], SIP_PAGE2_CONSTANT_SSRC)) { + if (p->rtp) { + ast_rtp_set_constantssrc(p->rtp); + } + if (p->vrtp) { + ast_rtp_set_constantssrc(p->vrtp); + } + } } else { /* No SDP in invite, call control session */ p->jointcapability = p->capability; ast_debug(2, "No SDP in Invite, third party call control\n"); @@ -21795,6 +21811,9 @@ static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask } else if (!strcasecmp(v->name, "t38pt_usertpsource")) { ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION); ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION); + } else if (!strcasecmp(v->name, "constantssrc")) { + ast_set_flag(&mask[1], SIP_PAGE2_CONSTANT_SSRC); + ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC); } else res = 0; @@ -23173,6 +23192,8 @@ static int reload_config(enum channelreloadreason reason) default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE; } else if (!strcasecmp(v->name, "matchexterniplocally")) { global_matchexterniplocally = ast_true(v->value); + } else if (!strcasecmp(v->name, "constantssrc")) { + ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_CONSTANT_SSRC); } else if (!strcasecmp(v->name, "session-timers")) { int i = (int) str2stmode(v->value); if (i < 0) { |