[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:all-tabpanel ]

dtryba updated ASTERISK-25268:
------------------------------

    Attachment: channel.c.diff

A diff againt asterisk 11-current (11.18.0). The indicate in set_format is needed for a change of SSRC to the A leg (from C) after completion of transfer. The indicate in ast_do_masquerade is needed to change SSRC to B (from C) during the attended transfer.

Tested with a patched Debian/jessie build (1:11.13.1~dfsg-2). None of the test calls fail with the patch where unpatched fails (ts going backwards)

In the latest git checkout of Asterisk channel.c is quite different. Will try to test current and patch if needed.

> 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: channel.c.diff, 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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