SIP is an application layer protocol for initiating, modifying, and terminating multimedia sessions over IP networks. It uses URIs to address endpoints and can be used for voice, video, instant messaging and other applications. SIP includes components like user agents, servers, and gateways. Common SIP messages are INVITE to create a session, ACK, BYE, and REGISTER. SDP is used to describe multimedia sessions and carry information like media formats and ports. Common codecs are G.711 for voice and H.264 for video.

1. 1
SIP Fundamentals
Chandra Prakash Singh

2. 2
Outline
• What is SIP
• SIP system components
• SIP messages and responses
• SIP call flows
• SDP basics/CODECs
• Questions and answers

3. 3
What’s SIP
• IETF Standard defined by RFC 3261
• “The Session Initiation Protocol (SIP) is an application-layer
control (signaling) protocol for creating, modifying and
terminating sessions with one or more participants.”
• Can be used for voice, video, instant messaging, gaming,
etc., etc., etc.
• Uses URIs for addressing – single communications identity
– mailto:chandra@plivo.com for email
– xmpp:29889_zentrunk@conf.hipchat.com for instant messaging
– sip:919980950111@zt.plivo.com for voice and video
• Username replaced by numbers for telephone applications

4. 4
Where’s SIP
Application
Transport
Network
Physical/Data Link Ethernet
IP
TCP UDP
RTSP SIP
SDP codecs
RTP DNS(SRV)

5. 5
SIP Components
• User Agents
– Clients – Make requests
– Servers – Accept requests
• Server types
– Redirect Server
– Proxy Server
– Registrar Server
– Location Server
• Gateways

6. 6
Terminating
User Agent
Originating
User Agent RTP
SIP SIP
B2BUA
SIP Peer to Peer !
Back-to-Back User Agent
Terminating
User Agent
Originating
User Agent
SIP
RTP

7. 7
SIP Methods
• INVITE Requests a session
• ACK Final response to the INVITE
• OPTIONS Ask for server capabilities
• CANCEL Cancels a pending request
• BYE Terminates a session
• REGISTER Sends user’s address to server

8. 8
SIP Responses
• 1XX Provisional 100 Trying
• 2XX Successful 200 OK
• 3XX Redirection 302 Moved Temporarily
• 4XX Client Error 404 Not Found
• 5XX Server Error 504 Server Time-out
• 6XX Global Failure 603 Decline

9. 9
SIP Flows - Basic
ACK
200 - OK
INVITE: sip:18.10.0.79
“Calls”
18.18.2.4
180 - Ringing Rings
200 - OK Answers
BYEHangs up
RTPTalking Talking
User
A
User
B

10. 10

11. 11

12. 12
SIP INVITE
INVITE sip:919980950111@phone.plivo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5062;branch=z9hG4bK.GlI7ViTR3;rport
From: <sip:inbound170412123914@phone.plivo.com>;tag=mpP3lkImP
To: sip:919980950111@phone.plivo.com
CSeq: 21 INVITE
Call-ID: qf8DhB2sA0
Max-Forwards: 70
Supported: replaces, outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 262
Contact: <sip:inbound170412123914@111.93.146.206:46219;transport=udp>;+sip.instance="<urn:uuid:51253a19-911f-4f24-a130-71a31e37f19d>"
User-Agent: Linphone/3.10.2 (belle-sip/1.5.0)
Proxy-Authorization: Digest realm="phone.plivo.com", nonce="W0X5jltF+GJsIaAdkgwQuhBG9MjKvaxz", username="inbound170412123914",
uri="sip:919980950111@phone.plivo.com", response="adeef6fff9bbebab570cfbbb32e85b29", cnonce="sWVndGo3ERdUNmXr", nc=00000001,
qop=auth

13. 13
Session Description Protocol
• IETF RFC 2327
• “SDP is intended for describing multimedia sessions for the
purposes of session announcement, session invitation, and
other forms of multimedia session initiation.”
• SDP includes:
– The type of media (video, audio, etc.)
– The transport protocol (RTP/UDP/IP, H.320, etc.)
– The format of the media (H.264 video, MPEG video,
etc.)
– Information to receive those media (addresses, ports,
formats and so on)

14. 14
SDP
v=0
o=inbound170412123914 886 3120 IN IP4 192.168.1.120
s=Talk
c=IN IP4 192.168.1.120
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=rtcp-fb:* trr-int 5000

15. 15
CODECs
• Audio
– G.711
• 8kHz sampling rate
• 64kbps
– G.729
• 8kHz sampling rate
• 8kbps
• Voice Activity Detection
• Video
– H.264
• MPEG-4
– H.263

16. 16
SIP Flows - Registration
200 - OK
REGISTER: sip
401 - Unauthorized
User
B sipphone
Registrar
REGISTER: (add
credentials)
DB
Location
sip:chandra@pliv
o.com
Contact IP

17. 17
SIP REGISTER
REGISTER sip:phone.plivo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5062;branch=z9hG4bK.yvX0~BYQ6;rport
From: <sip:inbound170412123914@phone.plivo.com>;tag=azPOuWU4j
To: sip:inbound170412123914@phone.plivo.com
CSeq: 20 REGISTER
Call-ID: O2NZOkRWpQ
Max-Forwards: 70
Supported: replaces, outbound
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact:
<sip:inbound170412123914@192.168.1.120:5062;transport=udp>;+sip.instance="<urn:uuid:51253
911f-4f24-a130-71a31e37f19d>"
Expires: 60
User-Agent: Linphone/3.10.2 (belle-sip/1.5.0)

18. 18
SIP REGISTER – 401 Response
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.120:5062;branch=z9hG4bK.yvX0~BYQ6;rport=46219;received=111.93.146.206
From: <sip:inbound170412123914@phone.plivo.com>;tag=azPOuWU4j
To: sip:inbound170412123914@phone.plivo.com;tag=378adae1ddc71986618d85886edb3f8b.6ead
CSeq: 20 REGISTER
Call-ID: O2NZOkRWpQ
WWW-Authenticate: Digest realm="phone.plivo.com",
nonce="W0X45ltF97pvAyUnhNauEQaleRnTOoJw"
Server: Plivo
Content-Length: 0

19. 19
SIP REGISTER with Credentials
REGISTER sip:phone.plivo.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5062;branch=z9hG4bK.FzY3F52HT;rport
From: <sip:inbound170412123914@phone.plivo.com>;tag=azPOuWU4j
To: sip:inbound170412123914@phone.plivo.com
CSeq: 21 REGISTER
Call-ID: O2NZOkRWpQ
Max-Forwards: 70
Supported: replaces, outbound
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact:
<sip:inbound170412123914@111.93.146.206:46219;transport=udp>;+sip.instance="<urn:uuid:512
9-911f-4f24-a130-71a31e37f19d>"
Expires: 60
User-Agent: Linphone/3.10.2 (belle-sip/1.5.0)
Authorization: Digest realm="phone.plivo.com", nonce="W0X45ltF97pvAyUnhNauEQaleRnTOoJw",
username="inbound170412123914", uri="sip:phone.plivo.com",
response="a1b449a5de1913f3616b6bcea1041552"

20. 20
SIP Standards
Just a sampling of IETF standards work…
IETF RFCs http://ietf.org/rfc.html
• RFC3261 Core SIP specification – obsoletes RFC2543
• RFC2327 SDP – Session Description Protocol
• RFC1889 RTP - Real-time Transport Protocol
• RFC2326 RTSP - Real-Time Streaming Protocol
• RFC3262 SIP PRACK method – reliability for 1XX messages
• RFC3263 Locating SIP servers – SRV and NAPTR
• RFC3264 Offer/answer model for SDP use with SIP

21. 21
SIP Standards (cont.)
• RFC3265 SIP event notification – SUBSCRIBE and NOTIFY
• RFC3266 IPv6 support in SDP
• RFC3311 SIP UPDATE method – eg. changing media
• RFC3325 Asserted identity in trusted networks
• RFC3361 Locating outbound SIP proxy with DHCP
• RFC3428 SIP extensions for Instant Messaging
• RFC3515 SIP REFER method – eg. call transfer
• RFC4474 Authenticated Identity Management
• SIMPLE IM/Presence - http://ietf.org/ids.by.wg/simple.html

22. 22
• “Hard phones”
• “Soft phones”
Soft and Hard SIP Clients

23. 23
Questions?

24. 24
Thank You