03 SS_SP004_E01_1 SIP_Protocol ZTE-44p.ppt

Network entities defined by SIP
Addressing solution defined by SIP
Commands defined by SIP
Communication mechanism defined by SIP
Simple call Scenario flow

1. NGN_SP004_E1
SIP Introduction

2. 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

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

4. RTCPRTP
SIP, H.323 and H.248
IP
H.248/Megaco
Call Control and Signaling Gateway control Media
H.225
Q.931
H.323
TCP
RAS
UDP
SIP
H.245
Video/
Audio
RTSP

5. 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.

6. 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

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

8. Redirect
Server
SIP Components – distributed architecture
Location
Server
Registrar
Server
User Agent
Proxy
Server
Gateway
PSTN
Proxy
Server
SIP
SIP
SIP
SIP
SIP
LDAP
LDAP

9. 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.

10. 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

11. 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

12. Basic SIP components (4/5)
 Network servers

Registrar server

accepts REGISTER requests

places the information it receives in those requests
into the location service

13. 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

14. SIP in ZXSS10 architecture
Core Packet NetworkCore Packet Network
ZXSS10 SS1A/B
Proxy server
Register server
Soft-phone
Video-phone
ZXSS10 SS1A/B
Proxy server
Register server

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

16. 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)

17. 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

18. SIP reply message
Message Function
1XX Temporary response
2XX Success
3XX Redirect
4XX Client error
5XX Server error
6XX Global error

19. SIP message format

20. SIP message format
Core Packet NetworkCore Packet Network
ZXSS10 SS1B
IP:202.202.21.1
Soft-phone
IP:202.202.41.8
SIP port: 5060
Number:6130000
Video-phone
IP:202.202.21.31
SIP port: 5060
Number:613000
1

21. SDP body
start lineINVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.202.41.8:5060
From: "iwf" <sip:[email protected]>;tag=aab7090044b2-195254e9
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 180
User-Agent: Cisco-SIP-IP-Phone/2
Accept: application/sdp
Contact: sip:[email protected]: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
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
Message head
SIP request message format

22. START SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 202.202.41.8:5060
To: <sip:[email protected]>;tag=caca1501-15112
From:
"iwf"<sip:[email protected]>;tag=aab7090044b2-
195254e9
Call-ID: 0009b7aa-124f0006-2050db78-
[email protected]
CSeq: 101 INVITE
User-Agent: ZTE Softswitch/1.0.0
Content-Length: 0
HEADER
SIP Reply message sample

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

24. SIP Call scenario analysis
Core Packet NetworkCore Packet Network
ZXSS10 SS1B
IP:10.41.6.1
Soft-phone
IP:10.66.74.136
SIP port: 5060
Number: #0* 109316
I704
IP:10.52.31.237
0755-26778086
PSTN Switch
sip
H.248

25. No.:12
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a
To: "0755526778086"<sip:[email protected]>
From: "#0*109316"<sip:#0*[email protected]>;tag=884a420a-
7062206315162668
Call-ID: [email protected]
CSeq: 23944 INVITE
Contact: <sip:#0*[email protected]: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
INVITE
SIP Call scenario analysis

26. No.:14
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a
To:"0755526778086"<sip:0755526778086@
10.41.6.1>;tag=a290601-31939
From:"#0*109316"<sip:#0*[email protected]>;ta
g=884a420a-7062206315162668
Call-ID: 072a13acfdc2669-
[email protected]
CSeq: 23944 INVITE
Contact: <sip:[email protected]>
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
INVITE
183 Ring
SIP Call scenario analysis

27. No.:15
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.66.74.136:5060;branch=z9hG4bK3af571e7266a
To:"0755526778086"<sip:[email protected].
6.1>;tag=a290601-31939
From:"#0*109316"<sip:#0*[email protected]>;tag
=884a420a-7062206315162668
Call-ID: 072a13acfdc2669-
[email protected]
CSeq: 23944 INVITE
Contact: <sip:[email protected]>
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
INVITE
183 Ring
200 OK
SIP Call scenario analysis

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

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

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

31. SIP in ZXSS10
ZXSS10 SS1A/B
Proxy server
Registrar server
ZXSS10 SS1A/B
Proxy server
Registrar server
Core Packet NetworkCore Packet Network
Soft-phone
Video-phone

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

33. 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
PSTN
Core Packet NetworkCore Packet Network
Video-phone
SG
MG
SS SS

34. SIP-T introduction
SIP-T means "SIP for Telephones", which is an expansion
of SIP protocol
PSTN
Core Packet NetworkCore Packet Network
Video-phone
SG
MG
SS SS
SIP-T

35. 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

36. SIP-T example
LS-1
Core Packet NetworkCore Packet Network
SG-1
MG-1
SS-1 SS-2
SIP-T
LS-2
SG-2
MG-2

37. 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

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

39. SIP-T sample analysis
LS-1 SS-2 LS-2SS-1
IAM
Invite (SDP+IAM)
IAM
ACM
180 (ACM)
ACM
ANM
200 (ANM+SDP)
Ack
ANM
conversation
REL
Bye (REL)
REL
RLC
RLC
200

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

41. Goals in generation of protocols
SIP H.323
Based on simple Internet Protocol models;
designed to meet converged (data, video,
voice) connectivity challenges
Based on the Telco model of
communications; evolved from the
telephone connectivity world
Standards established by the IETF Standards established by the ITU
Able to address the needs of a distributed
WAN infrastructure
Evolved from a LAN-centric view of the
Internet; disproportionate
suitable for carrier-class deployment
focus on telephone connectivity to the
exclusion of a rich data or video feature set

42. CAPABILITIES AND DESIGN INTENT
SIP H.323
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
IP is the carrier protocol for RTP (Real Time
Protocol) but the underlying behaviors of the
protocols are specified uniquely by H.323
Circuit reliability, or the lack thereof, is the
responsibility of the underlying network
infrastructure
Reliability is inherent in H.323 often introducing
unnecessary levels of service
SIP messages are transmitted as ASCII text
strings, consistent with email and web messages
(SMTP, POP, HTTP, etc.)
Evolved from a LAN-centric view of the Internet;
disproportionate
SIP allows architectural as well as
command/response extensions using well
documented methods
H.323 uses binary messaging
Efficient code implementation supporting easy of
embedding in minimum memory model devices
Complex, cumbersome code that is difficult to
implement in embedded systems
Architecture minimizes setup delay
As much as 7 or 8 seconds may be required to
negotiate circuit setup
Scalable, hierarchical addressing based on URL
syntax
Telco-like addressing with limitations on scalability

43. APPLICATION SERVICES
SIP H.323
Ability to ring more than 1 telephone end-point for
an incoming call (call 'forking') ie: office, home,
and cell phones all ring when a call is received.
No ability to fork calls
Individual user profile management
'Unified messaging'
Presence management
Media can be mixed in a single connection (voice,
data, streaming video)
No ability to mix media in a single call
Connection initiation through URL's that can be
embedded in web pages or other browser-based
devices
No ability to identify end-points with URL's
SIP allows seamless integration with other IP-
based protocols
H.323 capabilities are fixed and must be used in
the voice context of the PSTN
IP-based services allow easy interoperation with
various types of gateway and Internet devices
SS7 PSTN service model requires H.323 devices,
often with vendor-proprietary implementations