From 81ce60f6d442e9e681bdfa72bc3d0204ad1cc744 Mon Sep 17 00:00:00 2001 From: Alexander Traud Date: Thu, 24 Mar 2016 20:08:10 +0100 Subject: 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 --- 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 83d3ea0eb..91fb0b546 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -13530,7 +13530,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int } /* Finally our remaining audio/video codecs */ - for (x = 0; 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) { -- cgit v1.2.3