SIP Basic Training for VOIP testing

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