What is SIP and WebRTC?

SIP, the Session Initiation Protocol,  is the base protocol for the ACIP reference profile created by the EBU. It is the base for the CCM, Codec Call Monitor, software that we will publish in Project IRISbroadcast. During the conference “Broadcast Radio (R)Evolution 17” in Stockholm, Olle E. Johansson gave an introduction to the SIP protocol and adjacent protocols, like RTP, RTCP and WebRTC.  You can now download the slides from the presentation.

The SIP protocol is a protocol developed for realtime communication in general – from text messaging and chats to video calls and games. The base function is to enable users to find each other on the network, providing mobility and use of multiple devices. After finding each other, the users agree on a session both are capable of. In the world of broadcast radio contribution, the users are mostly audio codecs and the agreement is about which codecs to use in each direction of the transmission.

 

 


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