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Alexander Shulgin2014-03-27 11:15:38
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Alexander Shulgin, 2014-03-27 11:15:38

Where can I get an example / doc on implementing WebRTC SIP server side?

Good afternoon.
Task: to make a "SIP phone" in the browser to organize the work of the call center. SIP numbers are rented. Calls will be made to external phones (mobile). "SIP phone" will be part of a small custom CRM.

To solve the problem, I tried to start www.sipml5.org and jssip.net
(it doesn’t matter at all, the main thing is WebRTC).

I read a lot of things, but I just can’t defeat the server part, namely the websocket implementation for simpl5 or jssip. I can’t find any implementation examples, or at least a manual that at least roughly describes what is required.

As far as I understood, WebSocket will be something like a tunnel/transport?

I would be grateful for useful material on the server side SIP WebRTC topic.
ps: completely confused, it was always hard for me with telephony;)

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hbruser, 2014-04-06
@6pblcb

Actually, there are two options
1. SIP via Websocket
2. Custom signaling via Websocket and further SIP
In both cases, an intermediate server will be needed.
In the first case, this is a Stateful SIP proxy that can receive SIP via Websocket and then work with the provider via SIP/UDP.
You may of course be lucky and the VoIP provider holds the Websocket over SIP specification , but this is unlikely.
Those. look in the direction of Stateful SIP Proxy RFC7118.
The second option is not to use SIP on the browser side and use a server that can be accessed with simple commands via the JS API. For example call() login() answer() etc. Those. again, an intermediate server is needed, but this is no longer a Stateful SIP Proxy, but a Js translator in SIP. From working solutions known to me: Web Call Server .
In addition, there are a number of SAAS solutions with a RESTful API, where you do not need to host the server yourself.
By the way, in the first case, in addition to the Stateful SIP Proxy, a media server may be required, for example, when trying to connect to a PSTN phone via G.729. This codec is not supported in WebRTC and the connection will fail without such a server.

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