No Audio between sipml5 client and asterisk

[rawat he]

Hi all,

After following the link [http://code.google.com/p/sipml5/wiki/Asterisk] , user was registered happily to asterisk.

Made a call from sipml5 client to asterisk , call also gets connected. But there is no audio between them.

On getting connected to asterisk, I am playing standard prompts like below

        exten => 100,1,Answer()
        same => 100,n,Playback(vm-from)
        same => 100,n,Playback(demo-thanks)
        same => Hangup()

Asterisk log snippet :

    [2013-07-29 17:41:55.367-PM]     -- Executing [100@webrtc:2] Playback("SIP/100-00000000", "vm-from") in new stack
    [2013-07-29 17:41:55.370-PM]     -- <SIP/100-00000000> Playing 'vm-from.gsm' (language 'en')
    [2013-07-29 17:41:56.052-PM]     -- Executing [100@webrtc:3] Playback("SIP/100-00000000", "demo-thanks") in new stack
    [2013-07-29 17:41:56.055-PM]     -- <SIP/100-00000000> Playing 'demo-thanks.gsm' (language 'en')

Also, I have few Q's on the SDP sent by sipml5 client

a) Origin field has ip address as 0.0.0.0. Shouldn't it be my local IP Address or for privacy reasons it is set to 0.0.0.0?
                       
                            o=Mozilla-SIPUA-22.0 15781 1 IN IP4 0.0.0.0
                           
b) Connection data filed has different connection IP address? What is this IP? Where it is taking from?

                            c=IN IP4 202.52.56.214
                           
         Shouldn't it be my local machine IP Address? Or is it correct?

c) I think asterisk is sending audio to wrong IP. Log says
       
       [2013-07-29 17:41:54.859-PM] VERBOSE[2662][C-00000000] chan_sip.c: [2013-07-29 17:41:54.859-PM] Peer audio RTP is at port    202.52.56.214:51299

Please let me know what I am doing wrong? How to resolve this issue and proceed further.

If more information, logs, configuration files etc is needed to resolve it, please let me know.

Thank you
Rawat

Setup Info:

    FF 22, asterisk-11 (trunk), sipml5 (trunk), Ubuntu 12.04

Logs :

Asterisk CLI logs attached [ ast_cli_logs]
Asterisk FULL logs attached [ ast_full_logs]

[Mamadou DIOP]

Asterisk guys know claim flu support for WebRTC which means you have two solutions:

1) Use SIPML5 with latest Asterisk without any  patch. For any issue, report to Asterisk forum.

2) Or, use WebRTC2SIP between any Asterisk version (without any patch) and SIPML5. For any issue, report here.

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<ast_cli_logs.txt><ast_full_logs.txt>

[rawat he]

Hi Mamadou,

Many thanks for the reply.

Option 1 worked :). Asterisk (without any patch) with SIPML5.

Even dial-outs from asterisk are also working.

[Aimar]

Hi rawat he,

Could you tell me witch version of asterisk do you installed?
Do you use any pacth?

So to be sure you manage to use sipml5 and asterisk without webrtc2sip?

[rawat he]

Hi Aimar,

I used latest Asterisk 11 ( trunk ) version. No patch used.

Thanks,
Rawat

[Aimar]

Hi.

I'm trying to use Asterisk Latest Version - 11.5.0

For now sipml5 is not registered in asterisk.

In attachments, i send my asterisk confs and sipml5 log of the demo

Thanks for the help

Segunda-feira, 29 de Julho de 2013 13:38:22 UTC+1, rawat he escreveu:

[Mamadou DIOP]

You're trying to connect  to 'ws://192.168.0.108:5062' this is not valid.

To connect to Asterisk, the websocket url should be something like ws://ASTERISK_IP:ASTERISK_HTTP_PORT/ws. If Asterisk IP is 192.168.0.108 and http port 8088 (see http.conf) then, the url would be ws://192.168.0.108:8088/ws

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<extensions.conf><http.conf><rtp.conf><sip.conf><sipml5 demo log.txt><users.conf>

[Aimar]

Thanks for the quick answer.

now is given: ForbiddenThis is an asterisk error :(
could someone help?

"
[Aug  7 03:39:45] ERROR[26336]: chan_sip.c:16893 register_verify: 'WS' is not a valid transport for '1060'. we only use 'UDP'! ending call.
[Aug  7 03:39:45] NOTICE[26336]: chan_sip.c:27919 handle_request_register: Registration from '"1060"<sip:10...@192.168.0.108>' failed for '192.168.0.145:58150' - Device not configured to use this transport type

"

Segunda-feira, 29 de Julho de 2013 13:38:22 UTC+1, rawat he escreveu:

[Mamadou]

Please ask on Asterisk forum.

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[rawat he]

In sip.conf

    transport=udp.ws.wss ==>  transport=udp,ws,wss
    Redundant line ==> transport = udp ; Set the default transports. The order determines the primary default transport.

   sip.conf [general section]  has multiple transport configured. Asterisk will use the last configuration for transport.

In users.conf
    For peer 1060 [that needs to be registered] , add transport=udp,wss,ws