[Home]

Summary:ASTERISK-26523: chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
Reporter:Michael Keuter (mkeuter)Labels:
Date Opened:2016-10-29 10:10:07Date Closed:2016-11-04 13:04:47
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:13.12.0 13.12.1 Frequency of
Occurrence		Constant
Related
Issues:
Environment:AstLinux 1.2.7, Linux 3.2.80 64-bit, Asterisk 13.12.1, Beronet Berofix ISDN/SIP-gateway (FW: 3.0.12), ISDN BRI lineAttachments:( 0) Asterisk-13.12.1-rtp-timeout.pcap
( 1) asterisk-full-log.txt
( 2) dialpan-part.txt
( 3) sip-conf-part-updated.txt
Description:Asterisk 13.12.1 with chan_sip cuts incoming calls (coming from my Berofix ISDN/SIP-gateway) after 2 minutes. This does not happen with 13.11.2. This commit is related to that problem:

[http://git.asterisk.org/gitweb/?p=asterisk/asterisk.git;a=commit;h=93332cb1d0eea18021ea6538237297e627d6e2fc]

After reverting this commit the problem is fixed. "rtptimeout" in sip.conf is set to 120 secs (the default is commented out ("off")).

{code}
  -- Called SIP/28_yeal52p
  -- Connected line update to SIP/berofix-pstn-00000017 prevented.
  -- SIP/28_yeal52p-0000001c is ringing
  -- SIP/28_yeal52p-0000001c answered SIP/berofix-pstn-00000017
  -- Channel SIP/28_yeal52p-0000001c joined 'simple_bridge' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
  -- Channel SIP/berofix-pstn-00000017 joined 'simple_bridge' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>

2 minutes later:

[2016-10-29 12:51:08] NOTICE[25945]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call 'SIP/berofix-pstn-00000017' for lack of RTP activity in 121 seconds
[2016-10-29 12:51:08] NOTICE[25945]: chan_sip.c:29402 check_rtp_timeout: Disconnecting call 'SIP/28_yeal52p-0000001c' for lack of RTP activity in 121 seconds
  -- Channel SIP/28_yeal52p-0000001c left 'native_rtp' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
  -- Channel SIP/berofix-pstn-00000017 left 'native_rtp' basic-bridge <06e97373-9117-43b9-9551-3aa96b98afbc>
== Spawn extension (privat_standard, s, 17) exited non-zero on 'SIP/berofix-pstn-00000017'
{code}

Comments:By: Asterisk Team (asteriskteam) 2016-10-29 10:10:08.009-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Rusty Newton (rnewton) 2016-10-31 12:34:22.451-0500

Can you provide the following:

* sip.conf configuration for those channels.
* the dialplan involved for dialing.
* a packet capture of the calls that get disconnected.
* a "full" Asterisk log that correlates to the pcap. It should include notice, error, warning, debug and verbose log channel types. Turn verbose and debug up to 5 each.

We'd like to better understand exactly what is going on. As we don't currently have anyone else reporting this. Thanks!
By: Michael Keuter (mkeuter) 2016-10-31 14:14:02.999-0500

I attached the requested files.

192.168.2.72 is the ISDN-SIP-gateway
192.168.2.70 is the Asterisk PBX
192.168.2.14 is IP-phone
By: Yasin CANER (ycaner) 2016-11-01 10:00:18.669-0500

same problem happens on my asterisk 13.12.1 . if disabling rtptimeout parameter on sip.conf , it works fine.
By: Michael Keuter (mkeuter) 2016-11-01 12:50:18.408-0500

@Yasin: Yes, disabling rtptimeout works for me too, thanks for the hint. Do you also use ISDN/DAHDI or similar?
By: Michael Keuter (mkeuter) 2016-11-01 12:54:00.091-0500

Updated sip.conf with NAT settings. I also tried with "nat=no" for all channels, but the issue then still exists.
By: Michael Keuter (mkeuter) 2016-11-01 14:14:23.598-0500

To clarify this a bit: When "rtptimeout=120" is set in "sip.conf" the issue appears for me. When rtptimeout is commented out (the default setting) all works fine.
By: Yasin CANER (ycaner) 2016-11-01 15:05:47.744-0500

@Michael Keuter Nope , No dahdi or ISDN. calls between peers like 101 and 102 and nat is force_port ,comedia.
i tried to with one peer calls a answer and playback a record , and it works no problem. problem is about bridged channels and chan_rtp in my view.
maybe it can be test on chan_pjsip tomorrow.
By: Friendly Automation (friendly-automation) 2016-11-04 13:04:47.683-0500

Change 4303 merged by zuul:
Revert "chan_sip: Fix lastrtprx always updated"

[https://gerrit.asterisk.org/4303|https://gerrit.asterisk.org/4303]
By: Friendly Automation (friendly-automation) 2016-11-04 13:32:06.861-0500

Change 4302 merged by zuul:
Revert "chan_sip: Fix lastrtprx always updated"

[https://gerrit.asterisk.org/4302|https://gerrit.asterisk.org/4302]
By: Friendly Automation (friendly-automation) 2016-11-04 13:32:27.385-0500

Change 4301 merged by zuul:
Revert "chan_sip: Fix lastrtprx always updated"

[https://gerrit.asterisk.org/4301|https://gerrit.asterisk.org/4301]
By: Friendly Automation (friendly-automation) 2016-11-08 04:59:59.259-0600

Change 4338 merged by Joshua Colp:
Revert "chan_sip: Fix lastrtprx always updated"

[https://gerrit.asterisk.org/4338|https://gerrit.asterisk.org/4338]
By: Friendly Automation (friendly-automation) 2016-11-08 05:00:05.084-0600

Change 4339 merged by Joshua Colp:
Revert "chan_sip: Fix lastrtprx always updated"

[https://gerrit.asterisk.org/4339|https://gerrit.asterisk.org/4339]