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authorMark Spencer <markster@digium.com>2004-07-31 02:31:24 +0000
committerMark Spencer <markster@digium.com>2004-07-31 02:31:24 +0000
commita70d4443339e56062c85d432edebd9823a6786a7 (patch)
tree425448ac4c8726d99188630f912450a7e40a19ed
parent79a00a33f95eaf26fa600958e4be4d47c6463d55 (diff)
Improve debugging of RTP ports (bug #2131, heavily modified)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@3548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rwxr-xr-xchannels/chan_sip.c18
1 files changed, 13 insertions, 5 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 24777946e..756e11a9b 100755
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2682,17 +2682,25 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* RTP addresses and ports for audio and video */
sin.sin_family = AF_INET;
memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
+
/* Setup audio port number */
sin.sin_port = htons(portno);
- if (p->rtp && sin.sin_port)
+ if (p->rtp && sin.sin_port) {
ast_rtp_set_peer(p->rtp, &sin);
+ if (debug) {
+ ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ ast_log(LOG_DEBUG,"Peer audio RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ }
+ }
/* Setup video port number */
sin.sin_port = htons(vportno);
- if (p->vrtp && sin.sin_port)
+ if (p->vrtp && sin.sin_port) {
ast_rtp_set_peer(p->vrtp, &sin);
-
- if (sipdebug)
- ast_verbose("Peer RTP is at port %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ if (debug) {
+ ast_verbose("Peer video RTP is at port %s:%d\n", ast_inet_ntoa(iabuf,sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ ast_log(LOG_DEBUG,"Peer video RTP is at port %s:%d\n",ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port));
+ }
+ }
/* Next, scan through each "a=rtpmap:" line, noting each
* specified RTP payload type (with corresponding MIME subtype):