[Spce-user] sipwise 3.1 (CE) and websocket support

[Ashutosh Apte]

Hi Andreas,
    Thank you for your response. I can successfully register my webRTC
(using either sipML5 or jsSIP) client endpoint with sipProvider.

I've also created a SIP Peering Group to route calls to a legacy SIP
gateway. On dialing, the call hits the legacy gateway but I notice that the
SDP body in the INVITE message uses RTP/SAVF. This gateway does not support
sRTP. Is there a way to force the mediaproxy to use RTP only?

I've changed the parameter under Domains -> Preferences -> NAT and Media
Flow Control -> srtp_transcoding from "transparent" to "ForceRTP". The
kamailio proxy debug logs do show that the parameter was indeed changed but
it did not have any effect on the INVITE message.

What configuration parameter should I be changing to get this to work?

Thanks again for your help,
Ashu


Hi,

On 02/12/2014 01:44 AM, Ashutosh Apte wrote:
>* I tried to compare a kamailio configuration file at
*>* https://gist.github.com/jesusprubio/4066845
<https://gist.github.com/jesusprubio/4066845> with the one located
*>* at  /etc/kamailio/lb/kamailio.cfg but I cannot see a "listen" entry
*>* specifically for websockets. (I do see that it does load websocket.so)
*
There is no listen entry specifically for websockets, as it listens on
the standard SIP ports (5060 TCP for WS and 5061 TLS for WSS -> you have
to enable tls in config.yml for WSS).


>* What is the default port on which sip:provider listens for websocket
*>* requests?
*
See above. The WS URL for SIP is http://ip:5060/ws in v3.1

Andreas





On Tue, Feb 11, 2014 at 4:44 PM, Ashutosh Apte <apteashutosh at gmail.com>wrote:

> I've downloaded the sip:provider v3.1 (Virtualbox image) and have gone
> through the handbook to configure the system. But it seems that I've missed
> enabling websocket support.
>
> I can successfully register an endpoint using direct SIP but when I try to
> use jssip or sipml5, the connection establishment fails and the endpoint is
> never registered.
>
> I tried to compare a kamailio configuration file at
> https://gist.github.com/jesusprubio/4066845 with the one located
> at  /etc/kamailio/lb/kamailio.cfg but I cannot see a "listen" entry
> specifically for websockets. (I do see that it does load websocket.so)
>
> What is the default port on which sip:provider listens for websocket
> requests?
> (I tried pointing the javascript to port 80, 443, 1443, 5081 etc but they
> all fail with different response codes - XMLRPC Not allowed, 301 Moved
> Permanently etc)
>
> What change should I make to get the registrations working?
>
> Please help.
>
> Thanks,
> Ashutosh
>
>
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