From 1256aedf66b062c011959c422df1fa08e9f55522 Mon Sep 17 00:00:00 2001
From: Alexander Traud <pabstraud@compuserve.com>
Date: Mon, 26 Oct 2015 17:42:03 +0100
Subject: [PATCH] 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
---
 channels/chan_sip.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 8a7ca54540..f282966273 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13332,7 +13332,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 		}
 
 		/* 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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