[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway
bogdan at opensips.org
Thu Jun 29 04:43:48 EDT 2017
First, some questions regarding the desired topology:
1) the WS end-points should register in OpenSIPS or all the way
into Kamailio ?
2) also, the calls from the WS end-points should be all the time
sent to Kamailio ?
More or less, what I'm asking is : is OpenSIPS suppose to act as a
gateway from WS to UDP , but pass all the resulting traffic to Kamailio ?
OpenSIPS Founder and Developer
OpenSIPS Bootcamp 2017, Houston, US
On 06/28/2017 12:47 PM, Alex Megalokonomos wrote:
> We have the following scenario: our office call center is an Alcatel
> OmniPCX Office setup.
> This handles most of our needs and also provides 4 SIP extensions.
> These are provided by what appears to be a Kamailio SIP server v 3.2.2
> (no webrtc or websockets support)
> What we would like to do is set up an OpenSIPS instance to handle
> WebRTC and proxy everything to this Kamailio SIP server.
> The idea is to allow a web client (using sip js or something similar)
> to register / make / receive calls as one of the Kamailio extensions.
> I think half of the configuration is this :
> which I've already completed and indeed, clients can register to
> opensips and chat/make calls over websockets between them.
> How do I go about proxying registrations/invites/etc to the kamailio
> server instead?
> best regards
> Users mailing list
> Users at lists.opensips.org
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