[asterisk-users] Why is WebRTC treated differently from regular SIP in Asterisk

[Joshua C. Colp]

On Fri, Jun 23, 2023 at 11:38 PM TTT <lists at telium.io> wrote:

> I’m learning about WebRTC clients, and am wondering why Asterisk treats
> them differently from any other SIP client.
>
>
>
> The media (RTP) should be no different, so the only difference should be
> on the signaling side.  I noticed that the Asterisk wiki mentions the need
> for res_pjsip_transport_websocket, so does that mean Asterisk requires
> the signaling to occur over a websocket?
>
>
>
> If I used a SIPJS fork which places the signaling over UDP (eg
> https://github.com/cwysong85/sipjs-udp) will it just be a regular SIP
> client and I shouldn’t have to configure anything special in Asterisk, just
> regular PJSIP.
>

The signaling can go over whatever transport (UDP, Websocket, TCP, TLS).
Websockets are commonly used because as I stated in my other response it is
what the browser provides. From a media level WebRTC itself is different
because it uses additional standards than a regular SIP client. It does
ICE, STUN, TURN, DTLS-SRTP (which makes the SDP incompatible with non
DTLS-SRTP SDP), and others for media streams, packet loss, and more. Could
a normal SIP client use those? Yes. Do they? Usually no.

All of this isn't driven by Asterisk, but WebRTC.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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