[asterisk-bugs] [JIRA] (ASTERISK-25332) marker bit lost in outgoing stream when incoming stream has vad

Gergely Dömsödi (JIRA) noreply at issues.asterisk.org
Tue Aug 25 03:25:32 CDT 2015


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25332?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227349#comment-227349 ] 

Gergely Dömsödi edited comment on ASTERISK-25332 at 8/25/15 3:24 AM:
---------------------------------------------------------------------

I made a little [patch|^rtp_timestamp_skew_workaround.patch] to workaround to problem. It is not supposed to be production-ready (I lack detailed knowledge of the RTP stack to do that), but it seems to solve the issue for me. When applied, the marker bit is set every time in the outgoing stream when there is pause in the incoming audio.

Note that the signed-unsigned modification which was part of the debug patch is not used in this one.


was (Author: doome):
I made a little patch to workaround to problem. It is not supposed to be production-ready (I lack detailed knowledge of the RTP stack to do that), but it seems to solve the issue for me. When applied, the marker bit is set every time in the outgoing stream when there is pause in the incoming audio.

Note that the signed-unsigned modification which was part of the debug patch is not used in this one.

> marker bit lost in outgoing stream when incoming stream has vad
> ---------------------------------------------------------------
>
>                 Key: ASTERISK-25332
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25332
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_rtp_asterisk
>    Affects Versions: 11.18.0, 13.3.2, 13.4.0
>         Environment: Fedora 22
>            Reporter: Gergely Dömsödi
>            Assignee: Unassigned
>         Attachments: calls.pcap, in.pcap, messages.txt, out.pcap, pjsip.conf, rtp_timestamp_skew.patch, rtp_timestamp_skew_workaround.patch
>
>
> When asterisk receives an incoming call from a peer capable of VAD (rtp packets stop coming on silence, and start coming again on voice), in the outgoing leg of the call, the marker bit is not set on the first packet when the audio is restarted, so the called party is unable to play/sync audio properly.



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



More information about the asterisk-bugs mailing list