SIP is a signaling protocol used to establish, modify, and terminate multimedia sessions over IP networks. It was originally designed in 1996 to make voice and video calls over IP and has since expanded to support additional applications like instant messaging, file transfer, and presence information. SIP works by defining messages passed between endpoints to initiate, manage, and terminate calls or sessions. It is an application layer protocol that can operate over various transport protocols like TCP, UDP, and SCTP.

1. The Basics

2. signaling communications protocol
Controlling voice calls over (IP) networks
Controlling video calls over (IP) networks
Instant Messages ( IM )
Presence Management .

3. SIP was originally designed in 1996
The protocol was standardized as RFC 2543
In November 2000, SIP was accepted as a
3GPP signaling protocol and permanent
element of the IP Multimedia Subsystem (IMS)
architecture for IP-based streaming multimedia
services in cellular systems.
At 2013, the latest version of the specification
is RFC 3261 SIPv2 .

4. The protocol defines the messages that are
sent
between endpoints .
It govern establishment , modifying ,
termination and other essential elements of a
media streaming call.
SIP can be used for two-party (unicast) or
multiparty (multicast) sessions .

5. video conferencing
streaming multimedia distribution
instant messaging
presence information
file transfer
fax over IP
online games.

7. SIP is an application layer protocol
designed to be independent of the underlying
transport layer .
it can run on Transmission Control Protocol
(TCP) .
User Datagram Protocol (UDP)
Stream Control Transmission Protocol (SCTP).
It is a text-based protocol

8. For the transmission of media streams (voice,
video) SIP typically employs the Real-time
Transport Protocol (RTP)
OR
Secure Real-time Transport Protocol (SRTP).

9. Readable text-based format .
Uniform resource identifier (URI)
Normal transmission format
sip:username:password@host:port
which is called Address of record
If secure transmission is required
sips:username:password@host:port

10. User Agents (UA)
User Agent Client (UAC)
User Agent Server (UAS)
Servers
Proxy – forwards the request to the next hop
Registrar – accepts registrar requests
Redirect Server – finds alternative locations
Location Service – stores bindings.

11. Request/Response model
● UAC sends the request, UAS responds
● Requests starts with a request line
● INVITE sip: alice@example.com SIP/2.0
● Methods
● INVITE, ACK, BYE, CANCEL, REGISTER,
OPTIONS
● And many more..

15. SIP-address is used to locate and
communicate
with other users.
sip:alice@example.com
sips:alice@example.com
Each user typically have an Address of Record
contacted.

16. INVITE sip:bob@aboutsip.com SIP/2.0
To: <sip:bob@aboutsip.com>
From: <sip:alice@aboutsip.com>;tag=987lkajsdf89au
Call-ID: lkjasdf90989lkj
CSeq: 1 INVITE
Content-Type: application/sdp
Content-Length: 450
Record-Route:
<sip:192.168.1.52:5060;transport=tcp;lr>
Via: SIP/2.0/TCP 192.168.0.122:5060;branch=xxx;rport
...
Contact: <sip:192.168.0.122:3156;transport=TCP>

17. 200 OK Response
SIP/2.0 200 OK
To: <sip:bob@aboutsip.com>;tag=89uasdkfjoiu
From: <sip:alice@aboutsip.com>;tag=987lkajsdf89au
Call-ID: lkjasdf90989lkj
CSeq: 1 INVITE
Via: SIP/2.0/TCP 192.168.0.122:5060;branch=xxx;rport
...
Contact: <sip:192.168.0.22:3156;transport=TCP>
Record-Route:
<sip:192.168.1.52:5060;transport=tcp;lr>
Content-Type: application/sdp
Content-Length: 451

18. Headers carries important information about
e.g. routing or request and responses.
● The more important headers:
● To & From
● Via
● Contact
● Call-ID
● Route & Record-Route
CSeq

19. P-CSCF ( Proxy – CSCF ) is SIP proxy.
I- CSCF (Interrogating – CSCF ) is inbound
SIP proxy server located between HSS and S-
CSCF .
S- CSCF ( Serving – CSCF ) is SIP redirect
server .

22. SIP != VoIP
SIP can do VoIP but is so much more
SIP actually doesn't care about audio or
video at all
SIP helps you route messages through the
network.
SIP helps to locate your friends.