[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=227016#comment-227016 ] 

dtryba commented on ASTERISK-25268:
-----------------------------------

I have been fooling around with channel.c to see if I could set a marker on the A leg. 
Looking at set_format I'm really wondering why there is no marker being set (ast_indicate(chan, AST_CONTROL_SRCUPDATE)) if there is a need to change the format?

Adding the marker here results in the A leg being marked, I haven't been able to push this build to the production platform to see if this solved my problem. If the use of the marker is interpreted as http://www.cs.columbia.edu/~hgs/rtp/faq.html#marker mentions, it should solve the problem.

> 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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