[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:
------------------------------

    Description: 
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, it 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.

  was:
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 a A leg instead of 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, it 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.


> 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
>         Attachments: timestamp.pcapng.gz
>
>
> 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, it 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.



--
This message was sent by Atlassian JIRA
(v6.2#6252)