[asterisk-commits] chan sip: Do not send all codecs on INVITE. (asterisk[13])

[SVN commits to the Asterisk project]

Matt Jordan has submitted this change and it was merged.

Change subject: chan_sip: Do not send all codecs on INVITE.
......................................................................


chan_sip: Do not send all codecs on INVITE.

Since version 13, Asterisk sent all allowed codecs as callee, even when the
caller did not request/support them. In case of dynamic RTP payloads, this led
to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the
intersection between the requested and the supported codecs is send again.

ASTERISK-24543 #close

Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287
---
M channels/chan_sip.c
1 file changed, 1 insertion(+), 1 deletion(-)

Approvals:
  Anonymous Coward #1000019: Verified
  Matt Jordan: Looks good to me, approved
  Joshua Colp: Looks good to me, but someone else must approve



diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8a7ca54..f282966 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13332,7 +13332,7 @@
 		}
 
 		/* Finally our remaining audio/video codecs */
-		for (x = 0; x < ast_format_cap_count(p->caps); x++) {
+		for (x = 0; ast_test_flag(&p->flags[0], SIP_OUTGOING) && 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: merged
Gerrit-Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Matt Jordan <mjordan at digium.com>