[Asterisk-code-review] chan sip: Do not send all codecs on INVITE. Do not break on ... (asterisk[master])

[Alexander Traud]

Alexander Traud has uploaded a new change for review.

  https://gerrit.asterisk.org/2458

Change subject: chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers.
......................................................................

chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers.

Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those
codecs, which the caller did not request/support. That fix was not complete
because on the second Session Timer all codecs were sent again. Some VoIP/SIP
clients interpreted that complete codec-list as a change in the SIP session.
Because of that, Asterisk did not send the RTP audio via NAT anymore which
created a non-audio scenario after the second Session Timer fired.

ASTERISK-24543 #close

Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66
---
M channels/chan_sip.c
1 file changed, 1 insertion(+), 1 deletion(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/58/2458/1

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 09ab1a1..ffc2084 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13531,7 +13531,7 @@
 		}
 
 		/* Finally our remaining audio/video codecs */
-		for (x = 0; ast_test_flag(&p->flags[0], SIP_OUTGOING) && x < ast_format_cap_count(p->caps); x++) {
+		for (x = 0; p->outgoing_call && x < ast_format_cap_count(p->caps); x++) {
 			tmp_fmt = ast_format_cap_get_format(p->caps, x);
 
 			if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {

-- 
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Gerrit-MessageType: newchange
Gerrit-Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>