[asterisk-bugs] [JIRA] (ASTERISK-26050) chan_sip: WebRTC Audio + Video Negotiation Problem

[scgm11 (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26050?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

scgm11 updated ASTERISK-26050:
------------------------------

    Attachment: agent1001.log
                agent1004.log

1001 calls 1004

1001 has audio
1004 has no audio

> chan_sip: WebRTC Audio + Video Negotiation Problem
> --------------------------------------------------
>
>                 Key: ASTERISK-26050
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26050
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 13.9.0
>         Environment: WebRTC
> Ubuntu 16.04
>            Reporter: scgm11
>            Assignee: scgm11
>         Attachments: agent1001.log, agent1004.log, fulllog, fulllog.txt, full.txt, pcap.rar, sipdebug.txt, sip_integra.txt, sip.txt
>
>
> Im having an issue that I think is on sdp negotiation. using webrtc
> basically is one way audio, but this is not nat related, the problem is when I have 2 peers with audio and video capability, when both have this capability the part that answer the call answers with video even when on the initial INVITE theres is no video, but I think is using the capabilities of the peer instead of the offer. as the ice negotiation use different ports for audio and video seems that if I dont answer with video enable I dont have audio.
> Peer             User/ANR         Call ID          Format           Hold     Last Message    Expiry     Peer      
> 192.168.99.127   1003             bjkrs1u8ui42ot3  (alaw)           No       Rx: ACK                    1003      
> 192.168.99.249   (None)           244fb0d764f9f25  (nothing)        No       Rx: OPTIONS                <guest>   
> 192.168.99.124   1001             5a99af5106c996d  (alaw|vp8)       No       Tx: ACK                    1001   
> I attach a sip log from asterisk.
> Let me know if Im not clear enough.



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