[asterisk-bugs] [JIRA] (ASTERISK-21411) Re-invite without SDP causes one-way audio

[Michael L. Young (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-21411?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=205234#comment-205234 ] 

Michael L. Young edited comment on ASTERISK-21411 at 4/11/13 5:14 PM:
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In addition to the pcaps that you provided for 1.8, it would be helpful to provide a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Also, please include your SIP settings, general section and peer section.

In the 1.4 pcap I see RTP flowing but, in the 1.8 pcap I never see RTP flowing.  Have you checked the firewall on the 1.8 machine and made sure that the rtp ports are not being blocked?  I take it that 1.4 is on a different server than the 1.8 one?

If there was one-way audio, I would expect to see RTP at least from one side instead of nothing unless the idea is to have RTP flow between the endpoints and not through Asterisk.  A bit thrown off that there appears to be RTP going through on the 1.4 pcap if the reinvite is so that media flows directly between endpoints.  Perhaps I am not understanding your setup.
                
      was (Author: elguero):
    In addition to the pcaps that you provided for 1.8, it would be helpful to provide a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Also, please include your SIP settings, general section and peer section.

In the 1.4 pcap I see RTP flowing but, in the 1.8 pcap I never see RTP flowing.  Have you checked the firewall on the 1.8 machine and made sure that the rtp ports are not being blocked?  I take it that 1.4 is on a different server than the 1.8 one?

If there was one-way audio, I would expect to see RTP at least from one side instead of nothing.
                  
> Re-invite without SDP causes one-way audio
> ------------------------------------------
>
>                 Key: ASTERISK-21411
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-21411
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.21.0, 11.2.0
>         Environment: CentOS release 5.9 (Final)
>            Reporter: Olivier Lambert
>
> A re-invite without SDP is causing the voice to be established in one way. The exact same test done on asterisk 1.4.38 is not giving the issue, creating a two ways audio as it should. Pcaps of the two tests are available at http://www.misterlambert.net/pcaps.zip. Reinvite is located at frame 5127 for the 1.4 test while it is located at frame 7897 for the 1.8. The one way audio is caused by the missing RTP flow visible at frame 5166.

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