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authormmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2010-03-25 17:05:40 +0000
committermmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2010-03-25 17:05:40 +0000
commit1e4fccc0ce0d999e98d7c0c72046be7098b135c0 (patch)
tree8e1a590a10bb9a93f44baa767c8c3739847b7752
parent4dd5afa6b884e4969becb335557b1de3d2c5c487 (diff)
Fix crashes resulting from reading non-existent RTP streams.
Specifically, when using the CHANNEL dialplan function, it was possible to crash Asterisk by trying to get the rtpdest of a stream type that is not in use by the channel. This commit fixes that issue. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@254540 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r--channels/chan_sip.c19
1 files changed, 13 insertions, 6 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9878a2c31..c1f438fa5 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -19064,17 +19064,24 @@ static int acf_channel_read(struct ast_channel *chan, const char *funcname, char
if (!strcasecmp(args.param, "rtpdest")) {
struct sockaddr_in sin;
+ struct ast_rtp *stream;
if (ast_strlen_zero(args.type))
args.type = "audio";
- if (!strcasecmp(args.type, "audio"))
- ast_rtp_get_peer(p->rtp, &sin);
- else if (!strcasecmp(args.type, "video"))
- ast_rtp_get_peer(p->vrtp, &sin);
- else if (!strcasecmp(args.type, "text"))
- ast_rtp_get_peer(p->trtp, &sin);
+ if (!strcasecmp(args.type, "audio")) {
+ stream = p->rtp;
+ } else if (!strcasecmp(args.type, "video")) {
+ stream = p->vrtp;
+ } else if (!strcasecmp(args.type, "text")) {
+ stream = p->trtp;
+ }
+
+ if (!stream) {
+ return -1;
+ }
+ ast_rtp_get_peer(stream, &sin);
snprintf(buf, buflen, "%s:%d", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
} else if (!strcasecmp(args.param, "rtpqos")) {
struct ast_rtp_quality qos;