[asterisk-commits] oej: branch group/rana-early-media-is-gone-1.8 r389475 - in /team/group/rana-...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed May 22 06:35:28 CDT 2013
Author: oej
Date: Wed May 22 06:35:23 2013
New Revision: 389475
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=389475
Log:
Adding tons of debug to see what's going on
Modified:
team/group/rana-early-media-is-gone-1.8/channels/chan_sip.c
team/group/rana-early-media-is-gone-1.8/main/channel.c
team/group/rana-early-media-is-gone-1.8/main/rtp_engine.c
team/group/rana-early-media-is-gone-1.8/res/res_rtp_asterisk.c
Modified: team/group/rana-early-media-is-gone-1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/rana-early-media-is-gone-1.8/channels/chan_sip.c?view=diff&rev=389475&r1=389474&r2=389475
==============================================================================
--- team/group/rana-early-media-is-gone-1.8/channels/chan_sip.c (original)
+++ team/group/rana-early-media-is-gone-1.8/channels/chan_sip.c Wed May 22 06:35:23 2013
@@ -7564,10 +7564,13 @@
* UDPTL is created as needed in the lifetime of a dialog, its file
* descriptor is set in initialize_udptl */
if (i->rtp) {
+ ast_debug(3, "===> Setting file descriptors on channel \n");
ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
ast_rtp_instance_set_write_format(i->rtp, fmt);
ast_rtp_instance_set_read_format(i->rtp, fmt);
+ } else {
+ ast_debug(3, "===> NOT Setting file descriptors for RTP on channel \n");
}
if (needvideo && i->vrtp) {
ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
@@ -7830,12 +7833,18 @@
struct ast_frame *f;
if (!p->rtp) {
+
+ ast_debug(4, "===> Trying to read, but no RTP allocated - %s\n", ast->name);
/* We have no RTP allocated for this channel */
return &ast_null_frame;
}
+ if (ast->fdno != 8) {
+ ast_debug(5, "===> Read audio fd %-2.2d for %s\n", ast->fdno, ast->name);
+ }
switch(ast->fdno) {
case 0:
+ ast_debug(4, "===> Read audio for %s\n", ast->name);
f = ast_rtp_instance_read(p->rtp, 0); /* RTP Audio */
break;
case 1:
@@ -7957,6 +7966,7 @@
/* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast->_state != AST_STATE_UP) {
+ ast_debug(3, "====> SIP_READ Throwing away audio \n");
ast_frfree(fr);
fr = &ast_null_frame;
}
@@ -9722,6 +9732,7 @@
/* Setup audio address and port */
if (p->rtp) {
if (portno > 0) {
+ ast_debug(4, "====> Setting address and port to send to \n");
ast_sockaddr_set_port(sa, portno);
ast_rtp_instance_set_remote_address(p->rtp, sa);
if (debug) {
@@ -9735,6 +9746,7 @@
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
/* Ensure audio RTCP reads are enabled */
if (p->owner) {
+ ast_debug(3, "====> Setting RTCP read fd on channel \n");
ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
}
@@ -9760,6 +9772,7 @@
/* Silence RTCP while audio RTP is inactive */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
} else {
+ ast_debug(4, "====> Disabling RTP \n");
ast_rtp_instance_stop(p->rtp);
if (debug)
ast_verbose("Peer doesn't provide audio\n");
@@ -20881,6 +20894,7 @@
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
ast_rtp_instance_activate(p->rtp);
+ ast_debug(3, "==========> ACTIVATING RTP on this channel %s\n", p->callid);
} else {
/* Alcatel PBXs are known to send 183s with no SDP after sending
* a 100 Trying response. We're just going to treat this sort of thing
@@ -30097,6 +30111,7 @@
if (p->rtp) {
/* Prevent audio RTCP reads */
+ ast_debug(3, "====> DISABLING AUDIO RTCP READ on channel \n");
ast_channel_set_fd(chan, 1, -1);
/* Silence RTCP while audio RTP is inactive */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
@@ -30110,6 +30125,7 @@
* from a reinvite */
ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
/* Enable audio RTCP reads */
+ ast_debug(3, "====> Enabling AUDIO RTCP READ on channel \n");
ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(p->rtp, 1));
}
}
Modified: team/group/rana-early-media-is-gone-1.8/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/team/group/rana-early-media-is-gone-1.8/main/channel.c?view=diff&rev=389475&r1=389474&r2=389475
==============================================================================
--- team/group/rana-early-media-is-gone-1.8/main/channel.c (original)
+++ team/group/rana-early-media-is-gone-1.8/main/channel.c Wed May 22 06:35:23 2013
@@ -2641,6 +2641,7 @@
/* If this new fd is valid, add it to the epoll */
if (fd > -1) {
+ ast_debug(3, "===> Adding fd %d to channel %s for epoll listening \n", which, chan->name);
if (!aed && (!(aed = ast_calloc(1, sizeof(*aed)))))
return;
@@ -2655,7 +2656,10 @@
/* We don't have to keep around this epoll data structure now */
free(aed);
chan->epfd_data[which] = NULL;
- }
+ ast_debug(3, "===> Not Adding fd %d to channel %s for epoll listening \n", which , chan->name);
+ }
+#else
+ ast_debug(3, "===> No EPOLL: Adding fd %d to channel %s for listening \n", which, chan->name);
#endif
chan->fds[which] = fd;
return;
@@ -3989,10 +3993,12 @@
}
/* Clear the exception flag */
ast_clear_flag(chan, AST_FLAG_EXCEPTION);
- } else if (chan->tech && chan->tech->read)
+ } else if (chan->tech && chan->tech->read) {
+ ast_debug(4, "====> Reading on channel %s\n", chan->name);
f = chan->tech->read(chan);
- else
+ } else {
ast_log(LOG_WARNING, "No read routine on channel %s\n", chan->name);
+ }
}
/*
Modified: team/group/rana-early-media-is-gone-1.8/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/group/rana-early-media-is-gone-1.8/main/rtp_engine.c?view=diff&rev=389475&r1=389474&r2=389475
==============================================================================
--- team/group/rana-early-media-is-gone-1.8/main/rtp_engine.c (original)
+++ team/group/rana-early-media-is-gone-1.8/main/rtp_engine.c Wed May 22 06:35:23 2013
@@ -1728,6 +1728,7 @@
int ast_rtp_instance_activate(struct ast_rtp_instance *instance)
{
+ ast_debug(3, "========> Activating sir, just activating - Implemented ? %s \n", instance->engine->activate ? "yes" : "no" );
return instance->engine->activate ? instance->engine->activate(instance) : 0;
}
Modified: team/group/rana-early-media-is-gone-1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/group/rana-early-media-is-gone-1.8/res/res_rtp_asterisk.c?view=diff&rev=389475&r1=389474&r2=389475
==============================================================================
--- team/group/rana-early-media-is-gone-1.8/res/res_rtp_asterisk.c (original)
+++ team/group/rana-early-media-is-gone-1.8/res/res_rtp_asterisk.c Wed May 22 06:35:23 2013
@@ -523,10 +523,12 @@
struct ast_rtp *rtp = NULL;
int x, startplace;
+
/* Create a new RTP structure to hold all of our data */
if (!(rtp = ast_calloc(1, sizeof(*rtp)))) {
return -1;
}
+
/* Set default parameters on the newly created RTP structure */
rtp->ssrc = ast_random();
@@ -544,6 +546,8 @@
ast_debug(1, "Failed to create a new socket for RTP instance '%p'\n", instance);
ast_free(rtp);
return -1;
+ } else {
+ ast_debug(3, "==== > Created new socket for RTP stream \n");
}
/* Now actually find a free RTP port to use */
@@ -573,6 +577,8 @@
return -1;
}
}
+
+ ast_debug(3, "=====> New RTP instance created. Port %d\n", x);
/* Record any information we may need */
rtp->sched = sched;
@@ -2141,7 +2147,7 @@
if (rtp->sending_digit) {
ast_rtp_dtmf_continuation(instance);
}
-
+ ast_debug(4, "====> Going to read \n");
/* Actually read in the data from the socket */
if ((res = rtp_recvfrom(instance, rtp->rawdata + AST_FRIENDLY_OFFSET,
sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0,
@@ -2162,6 +2168,8 @@
/* Get fields and verify this is an RTP packet */
seqno = ntohl(rtpheader[0]);
+
+ ast_debug(3, "=====> read audio from socket Seqno %d\n", seqno);
ast_rtp_instance_get_remote_address(instance, &remote_address);
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