[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:comment-tabpanel&focusedCommentId=231214#comment-231214 ] 

scgm11 commented on ASTERISK-26050:
-----------------------------------

in the client I can see the invite like this:

INVITE sip:902bgr1l at 38aqm6okrur0.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.99.247:5060;branch=z9hG4bK12470f2e;rport
Max-Forwards: 70
From: "Agente3" <sip:... at 192.168.99.247>;tag=as01241992
To: <sip:902bgr1l at 38aqm6okrur0.invalid;transport=ws>
Contact: <sip:1003 at 192.168.99.247:5060;transport=WS>
Call-ID: 6cc70e4b63e333ba3c0e8d3c13be32f5 at 192.168.99.247:5060
CSeq: 102 INVITE
User-Agent: integraccs
Date: Mon, 23 May 2016 18:11:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1548

v=0
o=root 173926305 173926305 IN IP4 192.168.99.247
s=Asterisk PBX 13.9.0
c=IN IP4 192.168.99.247
b=CT:384
t=0 0
m=audio 11618 RTP/SAVPF 107 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:60
a=ice-ufrag:5aa0bc4a0a8c859b65cab5a867615742
a=ice-pwd:6b8ad0943e2964c56562a8b9148f5951
a=candidate:Hc0a863f7 1 UDP 2130706431 192.168.99.247 11618 typ host
a=candidate:Sba3609d7 1 UDP 1694498815 186.54.9.215 11618 typ srflx raddr 192.168.99.247 rport 11618
a=candidate:Hc0a863f7 2 UDP 2130706430 192.168.99.247 11619 typ host
a=candidate:Sba3609d7 2 UDP 1694498814 186.54.9.215 11619 typ srflx raddr 192.168.99.247 rport 11619
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 ED:C4:F0:B1:AB:97:6A:67:80:2A:BC:5E:B8:80:37:E3:94:61:DC:02:18:A1:97:79:FE:ED:38:6D:2C:0B:82:4D
a=sendrecv
m=video 15120 RTP/SAVPF 100
a=ice-ufrag:758d9a2a6e6701c82242fef41043a6d1
a=ice-pwd:7178441742ee7cd604bf36db4bda3888
a=candidate:Hc0a863f7 1 UDP 2130706431 192.168.99.247 15120 typ host
a=candidate:Sba3609d7 1 UDP 1694498815 186.54.9.215 15120 typ srflx raddr 192.168.99.247 rport 15120
a=candidate:Hc0a863f7 2 UDP 2130706430 192.168.99.247 15121 typ host
a=candidate:Sba3609d7 2 UDP 1694498814 186.54.9.215 15121 typ srflx raddr 192.168.99.247 rport 15121
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 ED:C4:F0:B1:AB:97:6A:67:80:2A:BC:5E:B8:80:37:E3:94:61:DC:02:18:A1:97:79:FE:ED:38:6D:2C:0B:82:4D
a=rtpmap:100 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv


this is asterisk invite has to comply with

https://tools.ietf.org/html/rfc3264 
https://tools.ietf.org/html/rfc4317 

a think this  "m=video 15120 RTP/SAVPF 100" shouldn't be there 


> 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: fulllog, fulllog.txt, full.txt, 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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