[asterisk-commits] trunk r33371 - in /trunk: channels/chan_sip.c
rtp.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Fri Jun 9 14:09:43 MST 2006
Author: oej
Date: Fri Jun 9 16:09:42 2006
New Revision: 33371
URL: http://svn.digium.com/view/asterisk?rev=33371&view=rev
Log:
- RTP debug message formatting
- Add rtcp report to SIP history
Modified:
trunk/channels/chan_sip.c
trunk/rtp.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=33371&r1=33370&r2=33371&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Jun 9 16:09:42 2006
@@ -2968,6 +2968,14 @@
if (!p->pendinginvite) {
/* Send a hangup */
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
+
+ /* Get RTCP quality before end of call */
+ if (recordhistory) {
+ if (p->rtp)
+ append_history(p, "RTCPaudio", "Quality:%s", ast_rtp_get_quality(p->rtp));
+ if (p->vrtp)
+ append_history(p, "RTCPvideo", "Quality:%s", ast_rtp_get_quality(p->vrtp));
+ }
} else {
/* Note we will need a BYE when this all settles out
but we can't send one while we have "INVITE" outstanding. */
@@ -3665,14 +3673,15 @@
if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */
p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
+
if (sin) {
p->sa = *sin;
if (ast_sip_ouraddrfor(&p->sa.sin_addr,&p->ourip))
p->ourip = __ourip;
- } else {
+ } else
p->ourip = __ourip;
- }
-
+
+ /* Copy global flags to this PVT at setup. */
ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
@@ -3682,6 +3691,7 @@
if (sip_methods[intended_method].need_rtp) {
p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
+ /* If the global videosupport flag is on, we always create a RTP interface for video */
if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) {
@@ -9551,7 +9561,7 @@
ast_log(LOG_DEBUG, " * SIP Call\n");
if (dialog->history)
AST_LIST_TRAVERSE(dialog->history, hist, list)
- ast_log(LOG_DEBUG, " %d. %s\n", ++x, hist->event);
+ ast_log(LOG_DEBUG, " %-3.3d. %s\n", ++x, hist->event);
if (!x)
ast_log(LOG_DEBUG, "Call '%s' has no history\n", dialog->callid);
ast_log(LOG_DEBUG, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid);
@@ -12626,6 +12636,14 @@
copy_request(&p->initreq, req);
check_via(p, req);
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
+
+ /* Get RTCP quality before end of call */
+ if (recordhistory) {
+ if (p->rtp)
+ append_history(p, "RTCPaudio", "Quality:%s", ast_rtp_get_quality(p->rtp));
+ if (p->vrtp)
+ append_history(p, "RTCPvideo", "Quality:%s", ast_rtp_get_quality(p->vrtp));
+ }
if (p->rtp) {
/* Immediately stop RTP */
ast_rtp_stop(p->rtp);
@@ -13700,6 +13718,8 @@
if (create_addr(p, host)) {
*cause = AST_CAUSE_UNREGISTERED;
+ if (option_debug > 2)
+ ast_log(LOG_DEBUG, "Cant create SIP call - target device not registred\n");
sip_destroy(p);
return NULL;
}
Modified: trunk/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/rtp.c?rev=33371&r1=33370&r2=33371&view=diff
==============================================================================
--- trunk/rtp.c (original)
+++ trunk/rtp.c Fri Jun 9 16:09:42 2006
@@ -428,7 +428,7 @@
attr = (struct stun_attr *)data;
if (ntohs(attr->len) > len) {
if (option_debug)
- ast_log(LOG_DEBUG, "Inconsistant Attribute (length %d exceeds remaining msg len %zd)\n", ntohs(attr->len), len);
+ ast_log(LOG_DEBUG, "Inconsistent Attribute (length %d exceeds remaining msg len %zd)\n", ntohs(attr->len), len);
break;
}
if (stun_process_attr(&st, attr)) {
@@ -1063,7 +1063,7 @@
rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
if (rtp_debug_test_addr(&sin))
- ast_verbose("Got RTP packet from %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
+ ast_verbose("Got RTP packet from %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
@@ -1086,7 +1086,7 @@
event_end >>= 24;
duration = ntohl(*((unsigned int *)(data)));
duration &= 0xFFFF;
- ast_verbose("Got rfc2833 RTP packet from %s:%d (type %d, seq %d, ts %d, len %d, mark %d, event %08x, end %d, duration %d) \n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
+ ast_verbose("Got RTP RFC2833 from %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
}
if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno);
@@ -2290,7 +2290,7 @@
}
if (rtp_debug_test_addr(&rtp->them))
- ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n",
+ ast_verbose("Sent RTP packet to %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen);
}
More information about the asterisk-commits
mailing list