WebRTC to SIP gateway questions

[sguti...@integraccs.com]

Hello,

At out company we develop software for Contact Centers, all webrtc agents using jssip and asterisk 13.

We are now building our next big release and have made many architectural changes, and one of those is fronting asterisk with "something" to manage the agents and their status and be able to communicate to asterisk.

The first approach is SIP Proxy and keep using jssip, but I would like to investigate if fronting with Janus could be a better way so I have some questions.

The idea is use the janus client JS library as the agent soft-phone.

I would like to know if is possible some things.

1) Exposing "presence"  online, offline, in use to "something"  in my case would like Redis, so I think we could contribute with a plugin in C to push this "presence" if available

2) "Registrar" so the agents login to janus via janus js client (I think that is out of the box)

3) Dialing a number to go to asterisk (I think this is the sip gateway) 

4) Is it possible to Dial from asterisk to an specific janus user? any examples on that?

5) As for transfers I use asterisk features I think it would work the same way (no refers)

6) basically every call from users go through asterisk and SIP endpoints (carriers) will get to asterisk and asterisk should be able to call the specific janus user right?

7) seems also a way to record video streams (an asterisk missing feature)

8) DTMF is available from the JS client right?

Couldn't find the differences between the 3 sip plugins, why are 3? I know sofia is the first one and suppose the most stable, but anywhere were I can read the reasons to add others?

So if you think all these are possible with Janus we will get deep on it and try to contribute to the project, using, testing and probably make some plugin.

Best Regards

[Lorenzo Miniero]

You may want to have a look at a few presentations I've done on SIP and Janus in the past:

Astricon:

https://www.slideshare.net/LorenzoMiniero/janusasterisk-astricon-2017

https://www.youtube.com/watch?v=IaPDufOzVek

OpenSIPS:

https://www.slideshare.net/LorenzoMiniero/janussip-opensips-2017

https://www.youtube.com/watch?v=anmyMC6Ovl8&t=25112s

These should give you a clearer understanding of what you can and cannot do, and how.

L.

[Sebastian Gutierrez]

Lorenzo,

Thanks for your feedback, I have seen the slides and presentations but my questions still remains, the main one is basically I always seen that all endpoints on Janus will register on Asterisk (one to one relationship?) so Im not offloading much from asterisk, or is there any way to make it act as a" SIP trunk" so Asterisk is not aware of the users and only one endpoint is managed between asterisk and Janus?

so trying to answer my questions based on those talks:

1) I will have to check the events, there should be the information

2) I´ve seen one to one, janus forwards to sip in asterisk, but I cant see if you can configure webrtc enpoints on juanus and not have more than one sip endpoint to asterisk

3) ok, janus will translate to the invite and everything for the call, I even see hold implemented as well I suppose this will work with asterisk

4) can’t tell from the talk due to all I have seen is one to one relationship between asterisk peers and janus users (is that how should be called?)

5) this should work

6) I need to understand Q2

7) ok

8) ok

Best regards

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[Lorenzo Miniero]

Janus just acts as an endpoint on behalf of WebRTC users, it's not a SIP infrastructure. You have to see it as a different SIP client. You offload Asterisk from all the WebRTC related aspects (ICE, DTLS encryption, maybe more). There's no SIP trunking since, again, it's not a proxy, it's a collection of SIP clients.

L.

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[Lorenzo Miniero]

When I say "Janus" here, I'm obviously talking of the SIP plugin. A different plugin may do things differently (e.g., provide the trunking you mention) but it's something you'd have to write yourself (we don't have any plans on that).

L.