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

Mon Aug 20 18:20:40 CDT 2012

Hi,
you need to build Asterisk with SRTP support...

*wget http://sourceforge.net/projects/srtp/files/latest/download -O
srtp-latest.tgz
tar -zxvf srtp-latest.tgz
./configure --prefix=/libsrtp
make && make install*

*And for Asterisk...*
*./configure --with-srtp=/libsrtp*
*
*
*this should work...*
*
*
*I did some changes to the sipml5 client and wanted to share this with you
guys... Actually only 2 simple changes...*
https://github.com/mailsvb/sipml5

*- The main config section has been splitted and made a little more
flexible, see *http://i45.tinypic.com/10x59o7.png
- Main call.html file has been renamed to .php and some code has been added
that will replace the "something.invalid" with the actual IP of your client
PC.

Currently I am able to register and at least make my softphone ring ;-) As
soon as I answer the outgoing call from sipml5 in the softclient, I get an
error in sipml5...

You can find my console output here http://pastebin.com/jdkXSMSD
I will continue investigating tomorrow...

best regards,
Sven

2012/8/20 Juan Castro <jcastro at instant.com.br>

> 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+Supportalong 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 --
> >>> New to Asterisk? Join us for a live introductory webinar every Thurs:
> >>>                http://www.asterisk.org/hello
> >>>
> >>> asterisk-users mailing list
> >>> To UNSUBSCRIBE or update options visit:
> >>>    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
> >>
> >> --
> >
> > 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
> > To UNSUBSCRIBE or update options visit:
> >    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
>
> --
> _____________________________________________________________________
> -- 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
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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