CODEWITHSUNDEEP
websocketwebrtcasterisksip
Umair
Umair

Reputation: 880

SIP communication with Web socket (Web RTC)

Sip (session initiation protocol) does not understand websocket so we need sip proxy which is basically a translator between sip and websocket. enter image description here

i am following this architecture for sip handshaking with web socket. I have few questions which sip proxy must be used to make audio and video call. and in the Gateway to SIP module i am using ASTERISK. how asterisk can be used for video call is there any codec available for video call? Please share some useful links. Your kind answers will be highly appreciated.

Upvotes: 1

Views: 7577

Answers (2)

jonpd
jonpd

Reputation: 356

Check out http://jssip.net. They provide a javascript API which uses SIP over WebSocket for client-side and they also have a SIP proxy and server (also works with Asterisk,Kamailio). They are the authors of RFC7118 "The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)".

Upvotes: 4

Dr. Alex Gouaillard
Dr. Alex Gouaillard

Reputation: 2128

  • that s only one way to do it. There are many ways.
  • you have to distinguish between the signaling path and the media path
  • on the signaling path, you have to choose a signalling protocol and corresponding transport protocol. A browser can use web socket for transport and sip for the protocol as far as signaling is concerned. On the legacy SIP side, you need SID over UDP, there is a need to change the transport of the signaling, not the protocol of the signaling.
  • On the media path, you have two problems, the encryption and the codec. The encryption is mandatory in webrtc and not in SIP. You need a B2BUA to make the transition between both words.
  • on the codec side, you either choose an overlapping codec between both words, or you have to transcode. The use of a media server seems mandatory here. If you have multiple parties in a conference, you will need to mix the audio and compose the video to send it to legacy SIP, in which case your media server should be an MCU.
  • Eventually, you also have a discovery and identity problem. During the original handshake, SIP is expecting a user ID and a domain (which is either a DNS entry or a fixed IP) while webRTC is using ICE. Here again, it is very likely that you need to use a B2BUA to bridge both world.

Asterisk/kamailio/freeswitch are likely to handle most of the above for the simple cases (1 to 1, audio). For anything complicated, you're on your own. You might want to look at respoke.io that was made by digium, the company behind asterisk.

Upvotes: 2

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