Summary:ASTERISK-23721: Calls to PJSIP endpoints with video enabled result in leaked RTP ports
Reporter:Marek Cervenka (cervajs)Labels:
Date Opened:2014-05-06 04:37:49Date Closed:2014-05-28 11:55:45
Status:Closed/CompleteComponents:Channels/chan_pjsip Resources/res_pjsip Resources/res_rtp_asterisk
Versions:SVN 12.2.0 Frequency of
Environment:centos6 64bit, pjproject rpm from digiumAttachments:( 0) pjsip.txt
Description:i tried performance test with sipp (sipp -r 30 -sn uac_pcap -l 200 -m 30000)
 sipp -> ast1.8 -> ast12(chan_pjsip)

the problem is in the ast12 box
Oh dear... we couldn't allocate a port for RTP instance
Unable to create RTP instance using RTP engine 'asterisk'

[root@voip ~]# netstat -lup|grep asterisk|wc -l

extensions.conf on ast12 box
exten => _X.,1,Answer()
exten => _X.,n,Playback(demo-instruct)
exten => _X.,n,hangup(16)


i can reproduce the problem

i tried last SVN - branch-12-r413282 - the same problem again
Comments:By: Matt Jordan (mjordan) 2014-05-06 10:34:18.017-0500

Please attach your {{pjsip.conf}}.

By: Marek Cervenka (cervajs) 2014-05-06 12:19:38.968-0500

[Edit by Rusty - removing inline file that should have been attached to the issue. I attached a config for use in reproduction]

By: Rusty Newton (rnewton) 2014-05-08 17:05:14.772-0500

Attaching pjsip.txt. Calls to the endpoint configured there will result in leaked RTP ports. The key appears to be having "allow = h264", as without that, RTP ports are not leaked.

I'm not sure exactly how RTP stuff works behind the scenes, but I did also test with chan_sip peers with videosupport=yes and video codecs enabled. The problem does not occur with chan_sip.

By: Richard Mudgett (rmudgett) 2014-05-27 18:24:25.304-0500

Patch up on reviewboard: