[asterisk-bugs] [JIRA] (ASTERISK-24846) Cancel Request Broken in chan_pjsip when it's used on Trunk with TCP transport

[Javier Riveros (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24846?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Javier Riveros  updated ASTERISK-24846:
---------------------------------------

    Comment: was deleted

(was: Hi Josua , i don't want to bother you with this again but i notice if i remove the transport=transport-tcp the outgoing INVITE is not TCP anymore even if the call was initiated with TCP on trunk become UDP ,i didn't notice this last Friday even is the origin is TCP.

{code}
[edgetrunk]
type=endpoint
;;transport=simpletrans-tcp
context=default
disallow=all
allow=ulaw
allow=alaw
allow=vp8
allow=h264
aors=edgetrunk
direct_media=no
rtp_symmetric=yes
media_address=XX.XX.XX.XX
force_rport=yes 
{code}

if left transport commented and  i add {{outbound_proxy=XX.XX.XX.XX\;transport=tcp}} transport become tcp but a new strange behavior appears after 30 seconds that  the call is stablished asterisk not pay attention to the (bye) coming from trunk if you hangup before 30 seconds hangup (bye) is proccesed by asterisk as it should.

Please let me know your comments or should i create another issue ?)

> Cancel Request Broken in chan_pjsip when it's used on Trunk with  TCP transport
> -------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24846
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24846
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 12.8.1, 13.2.0
>         Environment: Asterisk 13.2
> OS : Ubuntu 14.04
> Standart softphone 
>            Reporter: Javier Riveros 
>            Assignee: Joshua Colp
>         Attachments: Full_Ast_Debug_Cancel_TCP_trunk_pjsip, pjsip.conf.txt, Sip_capture_cancel.txt
>
>
> When asterisk request a CANCEL on a trunk it use different origin port used on INVITE it cause the trunk endpoint dont hangup the call because transaction doesn't exist.
> {code}
> - INVITE Generate by asterisk
> T 172.31.22.69:"56625" -> 172.31.16.57:5060 [A]
> INVITE sip:pbx.rarevase.com;transport=tcp SIP/2.0.
> Via: SIP/2.0/TCP 172.31.22.69:56625;rport;branch=z9hG4bKPj70f06330-08c0-48ae-8588-c3ab27839c68;alias.
> From: 'javier' <sip:javier at 172.31.22.69>;tag=a15f98b6-5f03-4b55-84f1-5f677759711a.
> To: <sip:c4e1a3 at pbx.rarevase.com>.
> Contact: <sip:425b1b25-1ed5-49d6-a561-52ed63c566dd at 172.31.22.69:56625>.
> Call-ID: 2aae6f80-f2c5-4c9b-94a0-37cfe99d2485.
> -CANCEL Send by asterisk
> ###
> T 172.31.22.69:"51875" -> 172.31.16.57:5060 [AP]
> CANCEL sip:pbx.rarevase.com;transport=tcp SIP/2.0.
> Via: SIP/2.0/TCP 172.31.22.69:51875;rport;branch=z9hG4bKPj70f06330-08c0-48ae-8588-c3ab27839c68;alias.
> From: 'javier' <sip:javier at 172.31.22.69>;tag=a15f98b6-5f03-4b55-84f1-5f677759711a.
> To: <sip:c4e1a3 at pbx.rarevase.com>.
> Call-ID: 2aae6f80-f2c5-4c9b-94a0-37cfe99d2485.
> -- Response from the gateway trunk
> T 172.31.16.57:5060 -> 172.31.22.69:"51875" [AP]
> SIP/2.0 481 Call/Transaction Does Not Exist.
> Via: SIP/2.0/TCP 172.31.22.69:51875;rport=51875;received=172.31.22.69;branch=z9hG4bKPj70f06330-08c0-48ae-8588-c3ab27839c68;alias.
> From: 'javier' <sip:javier at 172.31.22.69>;tag=a15f98b6-5f03-4b55-84f1-5f677759711a.
> To: <sip:c4e1a3 at pbx.rarevase.com>;tag=awAdaH.
> Call-ID: 2aae6f80-f2c5-4c9b-94a0-37cfe99d2485.
> {code} 
> If you see you can check that Call-ID is the same for all outbound transaction but the origin port on asterisk site is {{51875}} used for CANCEL is different it was using initially on INVITE {{56625}} so this causes that GATEWAY doesn't hangup the call in the other party.
> To reproduce this use a trunk using TCP transport  on chan_pjsip.
> i attached the full asterisk debug, sip_capture and pjsip.conf config. 



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