diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-05-17 16:11:26 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-05-17 16:11:26 +0000 |
| commit | d2683128536edad7b33b9d0faf1651f01d9f3455 (patch) | |
| tree | 50b88b8d50068ee82c90da7e9f3388b3944a8d80 /channels | |
| parent | 4c1702100630f429e6331b0537a584a02f92ea69 (diff) | |
Merged revisions 64754 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines
Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@64755 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 7e5499d44..732e1690e 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -17960,6 +17960,11 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc p->redircodecs = codecs; changed = 1; } + if ((p->capability & codecs) != p->capability) { + p->jointcapability &= codecs; + p->capability &= codecs; + changed = 1; + } if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) { if (chan->_state != AST_STATE_UP) { /* We are in early state */ if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) |