diff options
| author | Kinsey Moore <kmoore@digium.com> | 2011-11-11 20:15:16 +0000 |
|---|---|---|
| committer | Kinsey Moore <kmoore@digium.com> | 2011-11-11 20:15:16 +0000 |
| commit | a4365a8ae2b699d3a0de549b480800db9dd0ec85 (patch) | |
| tree | c041e5e95227ce18c885e989a198d2bb507313b4 /channels | |
| parent | e48cecc84871f0c17c02739f4db8af338a2a1d5d (diff) | |
Fix regression introduced by SDP fixups
If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails. Make sure capability is only touched if RTP is active. This
regression was introduced in R344385.
........
Merged revisions 344769 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 344770 from http://svn.asterisk.org/svn/asterisk/branches/10
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
| -rw-r--r-- | channels/chan_sip.c | 21 |
1 files changed, 12 insertions, 9 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 5896e6ed9..8d531c4a9 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -9290,15 +9290,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0)); } - /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since - they are acceptable */ - ast_format_cap_copy(p->jointcaps, newjointcapability); /* Our joint codec profile for this call */ - ast_format_cap_copy(p->peercaps, newpeercapability); /* The other sides capability in latest offer */ - p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */ - - if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */ - ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); - ast_format_cap_set(p->jointcaps, &tmp_fmt); + if (portno != -1 || vportno != -1 || tportno != -1) { + /* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since + they are acceptable */ + ast_format_cap_copy(p->jointcaps, newjointcapability); /* Our joint codec profile for this call */ + ast_format_cap_copy(p->peercaps, newpeercapability); /* The other sides capability in latest offer */ + p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */ + + /* respond with single most preferred joint codec, limiting the other side's choice */ + if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { + ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt); + ast_format_cap_set(p->jointcaps, &tmp_fmt); + } } /* Setup audio address and port */ |