In this video we’ll look at the protocols that make VoIP possible. These are SIP, SDP, and RTP.
SIP is used for call signalling. This means setting up calls, terminating them, and everything in between. The phone is usually the User Agent Client (UAC) while the PBX is the User Agent Server (UAS). When a phone comes online, it uses a SIP registration to inform the PBX of it’s IP address. If there’s an SBC in the path, it can pass on the SIP messages (making it a SIP proxy).
Much like HTTP, SIP uses a series of requests and responses, including headers and response codes.
Multiple SIP messages pass back and forth, usually between a phone and the PBX. The PBX is the intermediary between different phones.
To describe the media to be used in the phone call, SDP (Session Description Protocol) is used. This is separate to SIP, but works closely with it. SDP defines the codec to use, bandwidth requirements, and so on.
Once the phone call is set up, the actual voice data is transferred using RTP. This travels directly between phones, and not through the PBX. However, this will usually need to flow through an SBC if there is one in the path.
Overview of this video:
0:00 Introduction
0:18 Introduction to SIP
2:15 Requests and Responses
6:04 A SIP Transaction
8:12 SDP
10:37 RTP/RTCP
Patreon information: networkdirection.net/patreon/
Further Reading:
Understanding the SIP Via Header
andrewjprokop.wordpress.com/2...
The Main SIP Header Fields Explained
www.3cx.com/blog/voip-howto/s...
The Session Description Protocol
www.3cx.com/blog/voip-howto/s...
Understanding the Session Description Protocol
andrewjprokop.wordpress.com/2...
SIP - The Session Description Protocol
www.tutorialspoint.com/sessio...
RTP
www.voip-info.org/rtp
What is RTCP?
www.3cx.com/pbx/rtcp/
What is RTP?
www.3cx.com/pbx/rtp/
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