Video Conference: Asterisk with WebRTC

Hello Everyone,
I am new to the topic, so pardon my lack of understanding.

My company has an established SIP stack and we want to add Video Conferencing capability to it.
As per my current understanding, there are at least two ways how Video Calls/Conference can be established:

  1. Since SIP connection is format agnostic, it is possible to use SIP both for signaling and audio/video transport via RTP
  2. Another way is to use WebRTC in conjunction with a gateway to route media stream via SIP (does it make sense?)

Assuming I understand the options correctly, the questions are:

  1. Can I make a video call from WebRTC based solution to a SIP-only client via Asterisk?
  2. Do I need a WebRTC gateway to connect to the Asterisk server in its default configuration?
  3. What is the purpose of PjSIP and alike?

Thanks in advance for the clarification!

PJSIP is the framework that is used by asterisk to perform sip functions, and asterisk by itself can do video out of the box no need for extras, just use the latest version and follow one of those many tutorials out there, you can use a free softphone application called GS Wave available for both android and apple, in couple hours you could be doing video calls, another couple hours you could be also video calling from a web browser, peer to peer video is cool and important especially in company environment.

Now the most important question is : do you really need to set up an in house video conferencing solution while there are plenty of free options capable of doing hundreds of participants basically for pennies if not for free ?

I don’t think you can make outgoing calls to WebRTC [I think this is actually wrong, as I think WebRTC user agents can register, however, it is still the case that, for normal calls, the terminating side is a server, not a client], so the WebRTC end has to be the initial client, and the plain SIP end would normally be the initial server. However, given that, you can set up a call, as long as the pure SIP user agent supports the video codec that is mandated by WebRTC.

You can’t use video or do anything useful with Asterisk in a default configuration, you must configure it. However, once configured, you shouldn’t need a gateway, unless you need to transcode the video. (I interpret default configuration as meaning with extensions.conf.sample loaded; you can’t do anything at all without some configuration loaded.)

To originate and terminate SIP signalling and RTP media and interface them to the transport independent core of Asterisk, e.g. to convert a SIP BYE into an AST_CONTROL_HANGUP event. chan_dahdi provides a similar service for plain old analogue lines (and other types of circuit switched connections. For analogue lines it would, for example, translate the removal of the loop by an FXS device and generate AST_CONTROL_HANGUP to the core.

@khamlichi.khalil, @david551 thanks for the explanations! After your replies and additional reading, I understand the matter a little bit better…

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