[asterisk-bugs] [JIRA] (ASTERISK-25268) Neither a src change or marker after (attended) transfer

[dtryba (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25268?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227057#comment-227057 ] 

dtryba edited comment on ASTERISK-25268 at 7/27/15 7:36 AM:
------------------------------------------------------------

I tested my patch but it has no effect. The Ericcson needs a change of SSRC to pass the RTP with timestamps in the past. 

During testing the C to B RTP leg goes back in time with a marker, this isn't played back to B, after transfer is complete C to A gets a new SSRC and there is a full audio path both ways.

I'll have to splice in a test setup for easier debugging/patches.


was (Author: dtryba):
I tested my patch but it has no effect. The Ericcson needs a change of SSRC to pass the RTP with timestamps in the past.



> Neither a src change or marker after (attended) transfer
> --------------------------------------------------------
>
>                 Key: ASTERISK-25268
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25268
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Transfers
>    Affects Versions: 11.13.1
>         Environment: Debian/jessie
>            Reporter: dtryba
>            Assignee: Unassigned
>         Attachments: extensions.conf, sip.conf, timestamp.pcapng.gz, transfer.log
>
>
> On completing an attended transfer the original caller doesn't hear the transferee. Up until completion every party had a bidirectional audio.
> This is an almost duplicate of ASTERISK-15790 except here the problem is on the A leg instead of the C leg. The cause is exactly the same: a huge jump in the timestamp (going backwards) without either a change of srccid or marker in rtp.
> This behavior is similar to ASTERISK-23142, which occasionally happens without using the asterisk transfer feature (using SIP UA functions instead). The Vodafone Ericsson upstream refuses to forward RTP to the endpoints (apparently with reason).
> Frequently happens with Asterisk 11.13.1~dfsg-2, the changes to channel.c are minimal (and non related) in the latest 11 LTS compared to 11.13.



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