SIP, SDP, and RTP Work | Introduction to VoIP (Part 3)

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/
LET'S CONNECT
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#NetworkDirection
#networking
#voip

[@NetworkDirection]

Thanks for watching everyone! I hope you enjoyed this series. Patreon information: networkdirection.net/patreon/

[@bonguthandokhumalo7552]

The VERY BEST explanation I've seen on KZbin. Thank you so much Mr.

[@samson23]

Excellent video!! Probably the best explanation I've seen on you tube!

[@billyg.7032]

Excellent explanation. Thanks for sharing this knowledge!

[@NetworkDirection]

My pleasure!

[@user-xs8el7ec2d]

Thank you for your clear explaination!

[@keithsloan994]

I have been taking the 'SIP SCHOOL' SSCA course. The quality of your content is amazing as a free version :)

[@NetworkDirection]

Thanks Keith, that's great to heat :)

[@DragonDHunter]

@@NetworkDirection so is it application layer or session layer? What's the difference between origin and c fields in SDP

[@billmichae]

Nice pace and preso style.

[@NetworkDirection]

Thanks!

[@alisherafat8408]

Nice tutorial, thanks

[@user-dv1tn3rp9u]

Excelente explicación, muchas gracias por hacer que parezca fácil

[@NetworkDirection]

¡De nada! Me alegro de que haya ayudado.

[@nangialainoorzai3497]

First of all thank you so much! Can you share any source about voice networks (books etc )?

[@olekristianrannekleiv762]

Is SIP a part of the session layer? all I can find online say the application layer, even within the OSI model, but this to me makes no sense as SIP deals with initiating, maintaining and terminating real-time sessions.

[@NetworkDirection]

Yeah, I thnk so. I just think of anything that high up as 'upper layers'

[@nagytoth02]

Hey there. It was a great video with clear explanation, and thank you for making this video. Just a quick question: At 9:14 after the IPv4 address what does the "/127" mean? If it's an IPv4 address, it should be the subnet mask, that is below 32, right?

[@nagytoth02]

Actually, I looked this up, and it said it is a TTL Limit, so we can avoid a packet being circulated between routers, because when the TTL of the packet gets to 0 (after 127 hops), it gets dropped by the router.

[@maxwellchessdotcom6952]

Good video!

[@NetworkDirection]

Thanks!