Sip + audio bridge

[Mirko Brankovic]

Hi,

Is there a good way that sip plugin can be used to bridge in audio plugin or that is not possible without modifying plugins.

Or something can be done with rtp-fwd?

Thanks,

Mirko

[Lorenzo Miniero]

I don't see how rtp-fwd would help, as there wouldn't be any way for the SIP user to then inject their own media in the bridge. If you need to mix WebRTC and SIP users together, use something like Asterisk or Freeswitch to do audiomixing, and use the SIP plugin to have your WebRTC users join there.

L.

[Mirko Brankovic]

Thanks for reply,

It is more for multiple mixed webrtc users in audio bridge and someone from old pstn to join with sip to same audio bridge, in case user doesnt have browser in hands.

Or to send call out via sip plugin and maybe publish remote stream to bridge?

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[Mirko Brankovic]

D you have any thoughts on how to implement this one the best way?

[Mirko Brankovic]

Theoretically, freeswitch with verto module and webrtc implementation would probably work if there is no sdp conflict in negotiations.

[Lorenzo Miniero]

In our conferencing platform (the same we use to stream IETF meetings) we use Asterisk's ConfBridge as the audio conference bridge. Then SIP users can call Asterisk directly, and WebRTC users can join via the SIP plugin. For end users the effect is the same as the AudioBridge, the only difference is that the mixing is done elsewhere and that you have to handle presence (e.g., who joins/leaves) yourself.

L.

[Mirko Brankovic]

Oh ok. 

I was hoping to avoid 3rd party software in this picture, since there are enough plugins on Janus anyway.

Thanks,

Mirko

[Lorenzo Miniero]

In that case, you'll have to modify the AudioBridge plugin so that it also acts as a SIP server.

L.

[Mirko Brankovic]

Hi,

We managed to make it happen :)

SO with a custom FreeSwitch plugin (similar to mod_verto) that is connecting to Janus as client and can send x amount of calls into a specified audiobridge.

Since FreeSwitch is ICE, DTLS, and decode to Opus capable, now we can have N amount of WebRTC and PSTN/SIP callers in same room and having Janus as Media Server, audio mixer.

Regards,

Mirko  

Regards,

Mirko

[Lorenzo Miniero]

Cool, congratulations on this effort!

L.

[Mirko Brankovic]

Thanks for all your help.

I really hope something of it will end up as open source, but that depends on company policy ;)

[Peter T]

Hi Mirko,

I'm currently in the same boat and trying to figure out the best way to get sip clients (only audio) into a janus videoroom (in case they have no or poor internet access).

Can you give me some details about your implementation or maybe example code ?

Thanks in advance

L.

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Regards,

Mirko

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[Mirko Brankovic]

Hi Peter,

Unfortunately I can't share that part atm, dince it is company property ;p In case it goes open source, for what i'm cheering, I will let you know.

But basically, we already had a freeswitch module that converts sip messages and freeswitch states over http to signaling server. There is also mod_verto that is doing same thing over websockets..

So we added the re-invite options with opus and dtls to our module so that sdp can be exchanged with janus, in case call needs to go to conference room ;)

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