From e7da006562e77df839c6b5f26c7b69ec99539df4 Mon Sep 17 00:00:00 2001 From: Joshua Colp Date: Wed, 7 Mar 2007 17:55:11 +0000 Subject: [PATCH] Merged revisions 58240 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r58240 | file | 2007-03-07 12:52:58 -0500 (Wed, 07 Mar 2007) | 2 lines Ensure we have (or should have) at least one matching codec before attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@58241 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- channels/chan_sip.c | 2 +- main/rtp.c | 13 ++++++++++++- 2 files changed, 13 insertions(+), 2 deletions(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 3b040363fc..98a43b979b 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -17830,7 +17830,7 @@ static int sip_sipredirect(struct sip_pvt *p, const char *dest) static int sip_get_codec(struct ast_channel *chan) { struct sip_pvt *p = chan->tech_pvt; - return p->peercapability; + return p->peercapability ? p->peercapability : p->capability; } /*! \brief Send a poke to all known peers diff --git a/main/rtp.c b/main/rtp.c index acd139418a..1ff0a453d3 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -1537,7 +1537,7 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1) struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL; enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED, text_dest_res = AST_RTP_GET_FAILED; enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED, text_src_res = AST_RTP_GET_FAILED; - int srccodec, nat_active = 0; + int srccodec, destcodec, nat_active = 0; /* Lock channels */ ast_channel_lock(c0); @@ -1592,6 +1592,17 @@ int ast_rtp_early_bridge(struct ast_channel *c0, struct ast_channel *c1) srccodec = srcpr->get_codec(c1); else srccodec = 0; + if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec) + destcodec = destpr->get_codec(c0); + else + destcodec = 0; + /* Ensure we have at least one matching codec */ + if (!(srccodec & destcodec)) { + ast_channel_unlock(c0); + if (c1) + ast_channel_unlock(c1); + return 0; + } /* Consider empty media as non-existant */ if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr) srcp = NULL;