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chan_sip.c: Fix channel staging assertion failure.

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The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized.  The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.

* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.

* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.

* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential.  The callers must hold references to the passed in channel and
rtp objects.

* Eliminated sip_hangup() trying to get the bridge peer.  It is futile at
this point because the channel could never be in a bridge.

Review: https://reviewboard.asterisk.org/r/3431/
........

Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3
changes/97/197/1
Richard Mudgett 11 years ago
parent c6a2a513c2
commit d28af99e65
3 changed files with 83 additions and 45 deletions
Show Stats Download Patch File Download Diff File

84
channels/chan_sip.c
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@ -7248,40 +7248,29 @@ static int sip_hangup(struct ast_channel *ast)
}
if (!p->pendinginvite) {
RAII_VAR(struct ast_channel *, bridge, ast_channel_bridge_peer(oldowner), ast_channel_cleanup);
char quality_buf[AST_MAX_USER_FIELD], *quality;
/* We need to get the lock on bridge because ast_rtp_instance_set_stats_vars will attempt
* to lock the bridge. This may get hairy...
*/
while (bridge && ast_channel_trylock(bridge)) {
sip_pvt_unlock(p);
do {
CHANNEL_DEADLOCK_AVOIDANCE(oldowner);
} while (sip_pvt_trylock(p));
}
if (p->rtp || p->vrtp || p->trtp) {
ast_channel_stage_snapshot(oldowner);
}
char *quality;
char quality_buf[AST_MAX_USER_FIELD];
if (p->rtp) {
ast_rtp_instance_set_stats_vars(oldowner, p->rtp);
}
struct ast_rtp_instance *p_rtp;
if (bridge) {
struct sip_pvt *q = ast_channel_tech_pvt(bridge);
if (IS_SIP_TECH(ast_channel_tech(bridge)) && q && q->rtp) {
ast_rtp_instance_set_stats_vars(bridge, q->rtp);
}
ast_channel_unlock(bridge);
p_rtp = p->rtp;
ao2_ref(p_rtp, +1);
ast_channel_unlock(oldowner);
sip_pvt_unlock(p);
ast_rtp_instance_set_stats_vars(oldowner, p_rtp);
ao2_ref(p_rtp, -1);
ast_channel_lock(oldowner);
sip_pvt_lock(p);
}
/*
* The channel variables are set below just to get the AMI
* VarSet event because the channel is being hungup.
*/
if (p->rtp || p->vrtp || p->trtp) {
ast_channel_stage_snapshot(oldowner);
}
if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
if (p->do_history) {
append_history(p, "RTCPaudio", "Quality:%s", quality);
@ -26443,10 +26432,6 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
}
}
if ((p->rtp || p->vrtp || p->trtp) && p->owner) {
ast_channel_stage_snapshot(p->owner);
}
/* Get RTCP quality before end of call */
if (p->rtp) {
if (p->do_history) {
@ -26467,22 +26452,49 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
if (p->owner) {
RAII_VAR(struct ast_channel *, owner_relock, NULL, ast_channel_cleanup);
RAII_VAR(struct ast_channel *, owner_ref, NULL, ast_channel_cleanup);
struct ast_rtp_instance *p_rtp;
/* Grab a reference to p->owner to prevent it from going away */
owner_ref = ast_channel_ref(p->owner);
p_rtp = p->rtp;
ao2_ref(p_rtp, +1);
/* Established locking order here is bridge, channel, pvt
* and the bridge and channel will be locked during
* ast_rtp_instance_set_stats_vars */
ast_channel_unlock(owner_ref);
sip_pvt_unlock(p);
ast_rtp_instance_set_stats_vars(owner_ref, p->rtp);
if (peer_channel && IS_SIP_TECH(ast_channel_tech(peer_channel))) {
struct sip_pvt *q = ast_channel_tech_pvt(peer_channel);
if (q && q->rtp) {
ast_rtp_instance_set_stats_vars(peer_channel, q->rtp);
ast_rtp_instance_set_stats_vars(owner_ref, p_rtp);
ao2_ref(p_rtp, -1);
if (peer_channel) {
ast_channel_lock(peer_channel);
if (IS_SIP_TECH(ast_channel_tech(peer_channel))) {
struct sip_pvt *peer_pvt;
peer_pvt = ast_channel_tech_pvt(peer_channel);
if (peer_pvt) {
ao2_ref(peer_pvt, +1);
sip_pvt_lock(peer_pvt);
if (peer_pvt->rtp) {
struct ast_rtp_instance *peer_rtp;
peer_rtp = peer_pvt->rtp;
ao2_ref(peer_rtp, +1);
ast_channel_unlock(peer_channel);
sip_pvt_unlock(peer_pvt);
ast_rtp_instance_set_stats_vars(peer_channel, peer_rtp);
ao2_ref(peer_rtp, -1);
ast_channel_lock(peer_channel);
sip_pvt_lock(peer_pvt);
}
sip_pvt_unlock(peer_pvt);
ao2_ref(peer_pvt, -1);
}
}
ast_channel_unlock(peer_channel);
}
owner_relock = sip_pvt_lock_full(p);
@ -26511,10 +26523,6 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
}
}
if ((p->rtp || p->vrtp || p->trtp) && p->owner) {
ast_channel_stage_snapshot_done(p->owner);
}
stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
stop_session_timer(p); /* Stop Session-Timer */

2
include/asterisk/rtp_engine.h
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@ -1745,6 +1745,8 @@ int ast_rtp_instance_get_stats(struct ast_rtp_instance *instance, struct ast_rtp
* \param chan Channel to set the statistics on
* \param instance The RTP instance that statistics will be retrieved from
*
* \note Absolutely _NO_ channel locks should be held before calling this function.
*
* Example usage:
*
* \code

42
main/rtp_engine.c
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@ -1305,36 +1305,64 @@ char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_r
void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
{
char quality_buf[AST_MAX_USER_FIELD], *quality;
RAII_VAR(struct ast_channel *, bridge, ast_channel_bridge_peer(chan), ast_channel_cleanup);
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
char quality_buf[AST_MAX_USER_FIELD];
char *quality;
struct ast_channel *bridge = ast_channel_bridge_peer(chan);
ast_channel_lock(chan);
ast_channel_stage_snapshot(chan);
ast_channel_unlock(chan);
if (bridge) {
ast_channel_lock(bridge);
ast_channel_stage_snapshot(bridge);
ast_channel_unlock(bridge);
}
quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY,
quality_buf, sizeof(quality_buf));
if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
}
}
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
quality = ast_rtp_instance_get_quality(instance,
AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf));
if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
}
}
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
quality = ast_rtp_instance_get_quality(instance,
AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf));
if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
}
}
if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
quality = ast_rtp_instance_get_quality(instance,
AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf));
if (quality) {
pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
if (bridge) {
pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
}
}
ast_channel_lock(chan);
ast_channel_stage_snapshot_done(chan);
ast_channel_unlock(chan);
if (bridge) {
ast_channel_lock(bridge);
ast_channel_stage_snapshot_done(bridge);
ast_channel_unlock(bridge);
ast_channel_unref(bridge);
}
}
int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)

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