[asterisk-users] How to test Websocket support in SIP in Asterisk trunk?

[Juan Castro]

Put my sipml5 changes there. By the way, this is what happens when I
try to call a X-Lite extension from a sipml5 extension:

jcvmasterisk1*CLI>
  == Using SIP RTP CoS mark 5
[Aug 20 17:24:02] ERROR[22737][C-00000009]: chan_sip.c:32140
setup_srtp: No SRTP module loaded, can't setup SRTP session.
[Aug 20 17:24:02] WARNING[22737][C-00000009]: chan_sip.c:9974
process_sdp: Can't provide secure audio requested in SDP offer
jcvmasterisk1*CLI>

Trying to do the reverse... X-Lite stays in "Calling..." - in sipml5,
the right pane, with the local webcam thumbnailm, pops up, but no
"Answer" button. Only "Call" and "Hangup". Also, after a loooong time,
I get a ringing tone in X-Lite. And the webcam thing never goes away
in sipml5. What I get in the log is just this:

jcvmasterisk1*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [2010 at demo:1] Dial("SIP/2012-00000004", "SIP/2010")
in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/2010
jcvmasterisk1*CLI>

sipml5 to sipml5: "Not acceptable here". And the destination extension
is totally inert. Log:

jcvmasterisk1*CLI>
  == Using SIP RTP CoS mark 5
[Aug 20 17:30:58] ERROR[22747][C-0000000c]: chan_sip.c:32140
setup_srtp: No SRTP module loaded, can't setup SRTP session.
[Aug 20 17:30:58] WARNING[22747][C-0000000c]: chan_sip.c:9974
process_sdp: Can't provide secure audio requested in SDP offer
jcvmasterisk1*CLI>

Meh, same thing as simpl5-to-plain-SIP.

Juan

On Mon, Aug 20, 2012 at 4:00 PM, Andrew Latham <lathama at gmail.com> wrote:
> On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro <jcastro at instant.com.br> wrote:
>> Hoo-hah. It registers. Progress!
>>
>> Now... media. Or not.
>>
>> On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp <jcolp at digium.com> wrote:
>>> ----- Original Message -----
>>>> >
>>>> > The complete URL to use is http://<asterisk IP address or
>>>> > host>:8088/ws
>>>> >
>>>> > Note the /ws at the end. WebSocket support is only available there.
>>>> > Doing otherwise would have required core HTTP server changes,
>>>> > which I wanted to avoid. Depending on what you are testing with
>>>> > you may need to change it slightly to add that in.
>>>>
>>>> Well, I did the following changes in sipml5 and now I get a "Bad
>>>> Request" on REGISTER, instead of 404. Clearly, I'm still missing
>>>> something. Here are the changes I made:
>>>
>>> You are probably getting hit by a bug in Asterisk 11 that has been fixed.
>>>
>>> It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration.
>>>
>>> --
>>> Joshua Colp
>>> Digium, Inc. | Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>> Check us out at:  www.digium.com  & www.asterisk.org
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>>
>>
>> --
>> Juan Carlos Castro y Castro
>> Instant Solutions - Telefonia Gerando Resultado
>> http://www.instant.com.br
>> Principais capitais: 4063-6100
>> Demais regiões: (11)4063-6100
>>
>> --
>
> Juan
>
> Matt just opened
> https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document
> some of this.  Feel free to pipe in.
>
> --
> ~ Andrew "lathama" Latham lathama at gmail.com http://lathama.net ~
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>    http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
Juan Carlos Castro y Castro
Instant Solutions - Telefonia Gerando Resultado
http://www.instant.com.br
Principais capitais: 4063-6100
Demais regiões: (11)4063-6100