SS1_045_E2_1

SIP Protocol
Objectives

 Upon the completion of this chapter, you should

be able to understand ：
 Network entities defined by SIP
 Addressing solution defined by SIP
 Commands defined by SIP
 Communication mechanism defined by SIP
 Simple call Scenario flow
Outline

 SIP introduction
 SIP components
 SIP message structure
 Call scenario analysis
 SIP-T introduction
 SIP/H323 comparison
SIP, H.323 and H.248

Call Control and Signaling Gateway control Media

Video/
H.323 Audio

H.225
SIP H.248/Megaco
H.245 Q.931 RAS RTP RTCP RTSP

TCP UDP

IP
What is SIP?

“SIP: Session Initiation Protocol

SIP is a multimedia communication protocol established
by IETF. It is a text-based application-layer control protocol
independent of lower-layer protocols, designed to establish,
modify and terminate two-party or multi-party multimedia
sessions over the IP network.

”
What is SIP?

“ SIP was firstly researched by the MMUSIC IETF

workgroup in 1995 and recommended to be a standard by
IETF in 1999.
SIP uses HTTP and SMTP protocols.
SIP is still developing now. Relevant equipment vendors
and service providers have created an SIP forum:
www.sipforum.org

”
Outline

 SIP introduction
 SIP components
 SIP message structure
 Call scenario analysis
 SIP-T introduction
 SIP/H323 comparison
SIP Components – distributed architecture

LDAP SIP

LDAP
Location Redirect Registrar
Server Server SIP Server SIP

SIP SIP

PSTN

User Agent Gateway

Proxy Proxy
Server Server
Basic SIP components (1/5)
 User agents
 User agent client (UAC)

A user agent client is a logical entity that creates a new
request, and then uses the client transaction state machinery to
send it.
 User agent server (UAS)

A user agent server is a logical entity that generates a
response to a SIP request. The response accepts, rejects, or
redirects the request.
Basic SIP components (2/5)

 Network servers
 Redirect server

reduce the processing load on proxy servers

improve signaling path robustness

push routing information for a request back in a
response to the client
Basic SIP components (3/5)

 Network Servers
 Proxy server

An intermediary entity that acts as both a server and
a client for the purpose of making requests on behalf
of other clients

ensure that a request is sent to another entity
"closer" to the targeted user
Basic SIP components (4/5)

 Network servers
 Registrar server

accepts REGISTER requests

places the information it receives in those requests
into the location service
Basic SIP components (5/5)

 Network servers
 location server

is used by a SIP redirect or proxy server

store information about a callee's possible
location(s).

a list of bindings of address-of- record keys to zero
or more contact addresses

The bindings can be created and removed in many
way
SIP in ZXSS10 architecture

ZXSS10 SS1A/B ZXSS10 SS1A/B

Proxy server Proxy server
Register server Register server

Core Packet Network

Video-phone

Soft-phone
Outline

 SIP introduction
 SIP components
 SIP message structure
 Call scenario analysis
 SIP-T introduction
 SIP/H323 comparison
SIP Message – Request/Reply
 SIP components rely on the interaction of SIP
messages to communicate with each other, the
messaging mechanism is based on Client/Server,
and can be divided into two categories (request
and reply)
SIP Request

Message Function

INVITE Initialize a conversation

ACK Acknowledge the invite message

BYE End conversation

CANCEL Cancel the unsuccessful request

REGISTER Registration

OPTIONS Query the server capacity

INFO Pass the interaction contents of a certain call

SIP reply message

Message Function

1XX Temporary response

2XX Success

3XX Redirect

4XX Client error

5XX Server error

6XX Global error

SIP message format
SIP message format
ZXSS10 SS1B
IP:202.202.21.1

Core Packet Network

Soft-phone Video-phone
IP:202.202.21.31
IP:202.202.41.8
SIP port: 5060
SIP port: 5060
Number:6130000 Number:613000
1
SIP request message format
start line INVITE sip:6130001@202.202.21.1 SIP/2.0
Via: SIP/2.0/UDP 202.202.41.8:5060
From: "iwf" <sip:6136000@202.202.21.1>;tag=aab7090044b2-195254e9
To: <sip:6130001@202.202.21.1>
Call-ID: 0009b7aa-124f0006-2050db78-7fded6f5@202.202.41.8
CSeq: 101 INVITE
Message head Expires: 180
User-Agent: Cisco-SIP-IP-Phone/2
Accept: application/sdp
Contact: sip:6136000@202.202.41.8:5060
Content-Type: application/sdp
Content-Length: 224
v=0
o=CiscoSystemsSIP-IPPhone-UserAgent 17052 15931 IN IP4 202.202.41.8
s=SIP Call
c=IN IP4 202.202.41.8
t=0 0
SDP body m=audio 17522 RTP/AVP 0 8 18 101
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
SIP Reply message sample

START SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 202.202.41.8:5060
To: <sip:6130001@202.202.21.1>;tag=caca1501-15112
From:
HEADER "iwf"<sip:6136000@202.202.21.1>;tag=aab7090044b2-
195254e9
Call-ID: 0009b7aa-124f0006-2050db78-
7fded6f5@202.202.41.8
CSeq: 101 INVITE
User-Agent: ZTE Softswitch/1.0.0
Content-Length: 0
Outline

 SIP introduction
 SIP components
 SIP message structure
 Call scenario analysis
 SIP-T introduction
 SIP/H323 comparison
SIP Call scenario analysis

ZXSS10 SS1B
IP:10.41.6.1

sip H.248

I704
Core Packet Network
IP:10.52.31.237

Soft-phone PSTN Switch

IP:10.66.74.136
0755-26778086
SIP port: 5060
Number: #0* 109316
SIP Call scenario analysis

INVITE No.:12
INVITE sip:0755526778086@10.41.6.1 SIP/2.0
Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a
To: "0755526778086"<sip:0755526778086@10.41.6.1>
From: "#0*109316"<sip:#0*109316@10.41.6.1>;tag=884a420a-
7062206315162668
Call-ID: 072a13acfdc2669-884a420a@10.66.74.136
CSeq: 23944 INVITE
Contact: <sip:#0*109316@10.66.74.136:5060>
Max-Forwards: 70
User-Agent: ZTE MULTIMEDIA SIPPHONE/V1.0 04-01-10
Content-Type: application/sdp
Content-Length: 288

v=0
o=#0*109316 3507761179 3608424475 IN IP4 10.66.74.136
s=session SDP
c=IN IP4 10.66.74.136
t=0 0
m=audio 10000 RTP/AVP 0 4 8 18
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
m=video 10002 RTP/AVP 34
a=rtpmap:34 H263/90000
 SIP Call scenario analysis

No.:14
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a INVITE
To:"0755526778086"<sip:0755526778086@
10.41.6.1>;tag=a290601-31939 183 Ring
From:"#0*109316"<sip:#0*109316@10.41.6.1>;ta
g=884a420a-7062206315162668
Call-ID: 072a13acfdc2669-
884a420a@10.66.74.136
CSeq: 23944 INVITE
Contact: <sip:0755526778086@10.41.6.1>
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PR
ACK,UPDATE
User-Agent: ZTE Softswitch/1.0.0
Content-Type: application/sdp
Content-Length: 115

v=0
o=ZTE 32 32 IN IP4 10.41.6.1
s=phone-call
c=IN IP4 10.52.31.237
t=0 0
m=audio 4006 RTP/AVP 0
a=ptime:20
 SIP Call scenario analysis
No.:15
SIP/2.0 200 OK

Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a
To:"0755526778086"<sip:0755526778086@10.41. INVITE
6.1>;tag=a290601-31939
From:"#0*109316"<sip:#0*109316@10.41.6.1>;tag
183 Ring
=884a420a-7062206315162668
Call-ID: 072a13acfdc2669-
884a420a@10.66.74.136
CSeq: 23944 INVITE 200 OK
Contact: <sip:0755526778086@10.41.6.1>
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PR
ACK,UPDATE
Record-Route: <sip:10.41.6.1;lr>
User-Agent: ZTE Softswitch/1.0.0
Content-Type: application/sdp
Content-Length: 115

v=0
o=ZTE 32 32 IN IP4 10.41.6.1
s=phone-call
c=IN IP4 10.52.31.237
t=0 0
m=audio 4006 RTP/AVP 0
a=ptime:20
SIP Call scenario analysis

INVITE
No.:16
ACK sip:10.41.6.1;lr SIP/2.0
183 Ring
Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a
200 OK To: "0755526778086"<sip:0755526778086@10.41.6.1>
From:
ACK "#0*109316"<sip:#0*109316@10.41.6.1>;tag=884a420
a-7062206315162668
Call-ID: 072a13acfdc2669-884a420a@10.66.74.136
CSeq: 23944 ACK
Contact: <sip:#0*109316@10.66.74.136:5060>
Max-Forwards: 70
Route: <sip:0755526778086@10.41.6.1>
 SIP Call scenario analysis

No.:17
BYE sip:#0*109316@10.66.74.136:5060 INVITE
SIP/2.0
Via: SIP/2.0/UDP 183 Ring
10.41.6.1:5060;branch=776249e9.0
Via: SIP/2.0/UDP
10.52.31.237:5060;branch=4dcf5bd7
200 OK
To:
"#0*109316"<sip:#0*109316@10.41.6.1>;tag=
884a420a-7062206315162668 ACK
From:
"0755526778086"<sip:0755526778086@10.41.
6.1>;tag=a290601-31939
Call-ID: 072a13acfdc2669- conversation
884a420a@10.66.74.136
CSeq: 18927 BYE
Max-Forwards: 69
User-Agent: ZTE Softswitch/1.0.0 BYE
Content-Length: 0
SIP Call scenario analysis

INVITE
No.:18
183 Ring SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.41.6.1:5060;branch=776249e9.0
Via: SIP/2.0/UDP
200 OK
10.52.31.237:5060;branch=4dcf5bd7
To:
ACK "#0*109316"<sip:#0*109316@10.41.6.1>;ta
g=884a420a-7062206315162668
From:
"0755526778086"<sip:0755526778086@10.
conversation 41.6.1>;tag=a290601-31939
Call-ID: 072a13acfdc2669-
884a420a@10.66.74.136
CSeq: 18927 BYE
BYE Max-Forwards: 69

200 OK
SIP in ZXSS10

ZXSS10 SS1A/B
ZXSS10 SS1A/B
Proxy server
Proxy server
Registrar server
Registrar server

Core Packet Network

Video-phone

Soft-phone
Outline

 SIP introduction
 SIP components
 SIP message structure
 Call scenario analysis
 SIP-T introduction
 SIP/H323 comparison
SIP-T introduction
 Softswitch network is an integrated servce network, apart
from providing service for IAD, SIP subscribers, it also has
to consider to inherit the existing PSTN subscribers without
losing certain service properties
SS SS

Core Packet Network

SG

MG

PSTN
Video-phone
SIP-T introduction
 SIP-T means "SIP for Telephones", which is an
expansion of SIP protocol
SS SIP-T SS

Core Packet Network

SG

MG

PSTN
Video-phone
Essentials of SIP-T
 SIP-T is trying to provide a framework to
incorporate the traditional PSTN signals into SIP
message. SIP-T uses encapsulation and
translation to achieve the two essentials for SIP
network: transparency and routable
 In the inter-connecting node of PSTN and SIP
network, SS7 ISUP message has been
encapsulated into SIP message to make sure that
the service content will remain intact, while the
associating specific message has been extracted
and translated into corresponding SIP header to
make the routing possible
SIP-T example

SS-1 SIP-T SS-2

Core Packet Network

SG-1
SG-2
MG-1 MG-2

LS-1
LS-2
SIP-T sample analysis
 After the SS1 receives the ISUP message coming from
LS1, it will encapsulate and translate the package into SIP
form. Firstly, it will finish the header according to the
caller/callee information in ISUP, such as the From/TO
domain and Request-URI domain.
 For SS2, as the callee has been analyzed to be a PSTN
subscriber, the ss2 will extract the ISUP message from SIP
and route the call according to the local information
 As for the intermediate message, such as SUS or INR,
they have been encapsulated into Info. Message in SIP
SIP-T sample analysis

SIP ISUP

Invite IAM
180 Ring ACM
200 OK ANM
Bye/Cancel REL
SIP-T sample analysis

LS-1 SS-1 SS-2 LS-2

IAM
Invite (SDP+IAM)
IAM
ACM
180 (ACM)
ACM

200 (ANM+SDP) ANM

Ack
ANM

conversation

REL Bye (REL)

REL

RLC 200 RLC

Outline

 SIP introduction
 SIP components
 SIP message structure
 Call scenario analysis
 SIP-T introduction
 SIP/H323 comparison
Goals in generation of protocols

SIP H.323
Based on simple Internet
Based on the Telco model of
Protocol models; designed to
communications; evolved from
meet converged (data, video,
the telephone connectivity world
voice) connectivity challenges
Standards established by the
Standards established by the ITU
IETF
Able to address the needs of a Evolved from a LAN-centric view
distributed WAN infrastructure of the Internet; disproportionate
focus on telephone connectivity to
suitable for carrier-class
the exclusion of a rich data or
deployment
video feature set
CAPABILITIES AND DESIGN INTENT
SIP
Edge devices are identified in a standard
Internet manner (URLs, DNS lookup, MIME
encoding) and protocol interaction is
consistent with the general TCP/UDP/IP
world
Circuit reliability, or the lack thereof, is the
responsibility of the underlying network
infrastructure
SIP messages are transmitted as ASCII text
strings, consistent with email and web
messages (SMTP, POP, HTTP, etc.)
SIP allows architectural as well as
command/response extensions using well
documented methods
Efficient code implementation supporting
easy of embedding in minimum memory
model devices
Architecture minimizes setup delay
Scalable, hierarchical addressing based on
URL syntax
H.323
IP is the carrier protocol for RTP (Real Time
Protocol) but the underlying behaviors of the
protocols are specified uniquely by H.323
Reliability is inherent in H.323 often
introducing unnecessary levels of service
Evolved from a LAN-centric view of the
Internet; disproportionate
H.323 uses binary messaging
Complex, cumbersome code that is difficult
to implement in embedded systems
As much as 7 or 8 seconds may be required
to negotiate circuit setup
Telco-like addressing with limitations on
scalability
APPLICATION SERVICES
SIP H.323
Ability to ring more than 1 telephone end-
point for an incoming call (call 'forking') ie:
No ability to fork calls
office, home, and cell phones all ring when
a call is received.
Individual user profile management
'Unified messaging'
Presence management
Media can be mixed in a single connection
No ability to mix media in a single call
(voice, data, streaming video)
Connection initiation through URL's that can
be embedded in web pages or other No ability to identify end-points with URL's
browser-based devices
SIP allows seamless integration with other H.323 capabilities are fixed and must be
IP-based protocols used in the voice context of the PSTN
IP-based services allow easy interoperation SS7 PSTN service model requires H.323
with various types of gateway and Internet devices, often with vendor-proprietary
devices implementations