From e7da006562e77df839c6b5f26c7b69ec99539df4 Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
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;
--
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