[asterisk-bugs] [JIRA] (ASTERISK-22889) Segmentation fault when RTP going via ICE

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22889?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan updated ASTERISK-22889:
-----------------------------------

    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.

  was:
I have 2 servers with Asterisk 11.5, both have config like this:

[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

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


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


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.

    
> Segmentation fault when RTP going via ICE
> -----------------------------------------
>
>                 Key: ASTERISK-22889
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22889
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip, Resources/res_rtp_asterisk
>    Affects Versions: 11.2.0, 11.5.0, 11.6.0
>         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 @ 2.00GHz
>            Reporter: Alexandr Byvaltsev
>            Severity: Critical
>         Attachments: backtrace-13_09.txt, backtrace.txt
>
>
> 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.

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