CODEWITHSUNDEEP
google-chromewebrtcsipfirewall
johnnyD97
johnnyD97

Reputation: 23

Webrtc without websockets

I am trying to test sip capabilities of firewalls using webrtc. However I noticed using the servers needed for webrtc (stun turn websockets etc.) will give me a false positive in that it won’t catch nuanced issues with the ALGs. For reference this is being done from a chrome app so I can’t just run a native sip stack in the browser.

My Question: can I leverage webrtc to just send sip(invite, options, register) and not use any other methods that would get around the firewall?

Upvotes: 2

Views: 1420

Answers (2)

Kevin Roy
Kevin Roy

Reputation: 111

Actually there might be a way around that.

Use the signalling server to to do any sort of preconfigurations you might want to do before setting up the peer connection. This would allow you to specify codecs and resolution of the feed as a SessionDescription before hand or even check if the other peer is capable of WebRTC or not.

I'd recommend Socket.io =D

Upvotes: 0

xdumaine
xdumaine

Reputation: 10329

Your question doesn't make sense because WebRTC doesn't use SIP - SIP is a signaling protocol, and WebRTC doesn't do signaling. What that means is that SIP can be used to establish a WebRTC connection, but they are mutually exclusive.

SIP is sent over a data connection, like a hard line from a phone to a PBX or a websocket from a browser to a server.

It is possible to set up a WebRTC connection using out of band mechanisms, but then that wouldn't be SIP.

Upvotes: 3

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