Is it possible to implement conference call between two sip handle and a same sip client that is
Sip plugin 1 => sip client <= Sip plugin 2
without making conference call on the client side that is without merging the calls on the client side
client should only recieve call from single number
The SIP plugin won’t mix audio, so you either need to use a separate SIP conference briddge, or keep multiple calls up and running at the same time for the conferencing part.
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@lorenzo basically our use case is something like this
Agent(janus sip plugin) makes a voip call to client’s mobile phone number and we want to be able to add third agent(janus sip plugin) to this ongoing call without letting client know that a new call was made which then client have to merge on his device to make it a conference call, we want that to happen on janus itself
Or otherwise have an ability to interlink media from sip and audiobridge plugin i believe this interlinking feature would make such use cases easy to implement making janus live upto its name (the general purpose webrtc server) along with sip stack
I don’t know how complicated it is to implement such thing but i would love to hear your views on this.
We really want to stick with janus for everything atleast related to webrtc and sip
Thanks
S.T
The SIP plugin doesn’t have mixing features. Getting AudioBridge and SIP plugins to work together is not easy but can be done: Bridging AudioBridge and SIP with Drachtio | Meetecho Blog