Re: [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  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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Re: [asterisk-users] WebRTC signaling

[Joshua C. Colp]

On Fri, Jun 23, 2023 at 11:19 PM TTT  wrote:

> I’m looking at using Asterisk 20 with WebRTC clients (sipjs).  I know the
> media runs over TCP, but what about the signaling?
>
>
Media doesn't generally go over TCP, it goes over UDP.


>
>
> I read something about signaling over UDP was proposed as part of a webrtc
> standard, but can’t find if that was ever ratified or if Asterisk can even
> use UDP for the signaling instead of TCP for the signaling.
>
>
>
> Does encryption of the signaling (SIPS) change anything?
>

WebRTC doesn't define signaling. SIP is an option, and the browser provides
websockets for its transport. It's all in what the browser supports.

-- 
Joshua C. Colp
Asterisk Project Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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