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Summary:ASTERISK-22889: Segmentation fault when RTP going via ICE
Reporter:Alexandr Byvaltsev (alfox)Labels:
Date Opened:2013-11-22 04:05:25.000-0600Date Closed:2017-12-18 11:10:48.000-0600
Priority:CriticalRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General Resources/res_rtp_asterisk
Versions:11.2.0 11.5.0 11.6.0 Frequency of
Occurrence
Related
Issues:
duplicatesASTERISK-22937 Asterisk crashes with pj_NO_MEMORY_EXCEPTION exception
is related toASTERISK-20762 Asterisk Crash, assertion failed, in res_rtp_asterisk thread (ice_worker_thread)
is related toASTERISK-21696 Assertion error results in crash in pjproject's ICE worker thread
is related toASTERISK-22938 Asterisk Crashes with assert fail for 'ype <= PJ_ICE_CAND_TYPE_RELAYED'
is related toASTERISK-23017 Crash on inbound calls using WebRTC config with ICE Servers -signal 6 abort, while in ice_worker_thread
Environment:Debian 7.2 on Intel(R) Core(TM) i5-3330 CPU @ 3.00GHz, Debian 7.0 on Intel(R) Xeon(R) CPU E5-2620 0 @ 2.00GHzAttachments:( 0) backtrace_27.11.2013.txt
( 1) backtrace.txt
( 2) backtrace-13_09.txt
( 3) rtp.conf
( 4) sip.conf
Description:I have 2 servers with Asterisk 11.5, both have config like this:

{noformat}
[trunk_74]
type=friend
host=192.168.66.99
port=5060
disallow=all
allow=alaw
context=trunk_74_in
t38pt_udptl=no
call-limit=0
qualify=no
canreinvite=no
dtmfmode=inband
nat=force_rport,comedia
avpf=yes
icesupport=yes
videosupport=no
directmedia=no
encryption=yes
{noformat}

when i make 200 calls at one time from one asterisk to another, in dialplan

{noformat}
007777 => {
             Answer();
             Wait(10);
             Hangup();
};
{noformat}

i got Segmentation fault in 1-5 minutes, only when i used icesupport=yes.

Same problem i have when sip users connect to Asterisk from browser Chrome over WEBRTC.
Comments:By: Matt Jordan (mjordan) 2013-11-26 12:41:11.347-0600

The fact that this is crashing during a free in the PJSIP memory pool is troubling, as that's indicative of a memory corruption somewhere.

Can you attach your full {{sip.conf}} and {{rtp.conf}}?
By: Alexandr Byvaltsev (alfox) 2013-11-27 00:11:14.442-0600

yes, sip.conf and rtp.conf in attachments
By: Joshua C. Colp (jcolp) 2017-12-18 11:10:49.044-0600

I'm suspending this issue since we now use PJSIP upstream and work has gone in there to fix these kind of problems. We also haven't seen any other reports in recent versions.