[asterisk-bugs] [JIRA] (ASTERISK-26992) chan_sip: Bell Canada Interop on Asterisk 13.15 fails due to no RTP on SDP sendrecv

David Brillert (JIRA) noreply at issues.asterisk.org
Fri May 4 15:28:56 CDT 2018


     [ https://issues.asterisk.org/jira/browse/ASTERISK-26992?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

David Brillert closed ASTERISK-26992.
-------------------------------------

    Resolution: Workaround Available

> chan_sip: Bell Canada Interop on Asterisk 13.15 fails due to no RTP on SDP sendrecv
> -----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26992
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26992
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.15.0
>         Environment: Asterisk 13.15, CentOS6 x64
>            Reporter: David Brillert
>         Attachments: fail with rtp.pcapng, pass with rtp.pcapng
>
>
> I have been working to complete Bell Canada interop testing with Bell Canada and Asterisk 13.15 but the interop does not pass due to outgoing call setup failure.  When using 11.25 the test case does not fail.  Since 11.25 is no longer maintained I am trying to complete the interop with *13
> I ran tests and pcaps using both versions to diagnose where the RTP is failing.
> Because the 183 session in progress message SDP has sendrecv Bell expects RTP from Asterisk.
> The passed test case is using Asterisk 11.25 and Asterisk is sending RTP in response to sendrecv. In the Asterisk 13.15 failure Asterisk does not send RTP after the sendrecv.
> Two traces are attached:
> 1. Asterisk 11.25 'pass with rtp.pcapng' where RTP is established after the sendrecv and the call completes OK.
> 2. Asterisk 13.15 'fail with rtp.pcapng' where RTP is not established after the sendrecv and the call fails.
> I have tried multiple setting in sip.conf including progressinband=yes and no and never without any improvement.
> sip.conf
> {noformat}
> transport       =  udp
> icesupport      =  no
> nat             =  no
> directmedia     =  yes
> insecure        =  port,invite
> disallowed_methods =  UPDATE
> session-timers  =  accept
> session-expires =  1800
> session-minse   =  90
> session-refresher =  uac
> encryption      =  no
> qualify         =  yes
> qualifyfreq     =  60
> disallow        =  all
> allow           =  ulaw
> {noformat}



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