Transcript SIP ,SIP-T and SIP-I Protocol 課程 : 新世代網路 教授 : 連耀南

SIP ,SIP-T and SIP-I
Protocol
課程 : 新世代網路
教授 : 連耀南
學生姓名 鄭旭鈞
學號 :93971004
Outline
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SIP
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SIP introduction
SIP Architecture
 Components of SIP
 SIP operation mode
 SIP message structure
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PSTN and SIP interworking
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SIP-T
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What’s SIP-T
Architecture
Translation of SIP-T
Encapsulation of ISUP in SIP-T
SIP-I
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Interworking architecture
Protocol overview
Specification of SIP-I
TRQ. 2815
Q.1912.5
Summary
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SIP Introduction
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Open, simple, extensible, and lightweight protocol
Design for IP Networks
Text-encoded protocol based on elements from
the HTTP and SMTP.
Same protocol used between services and call
control entities Text-based encoding
Easier to integrate with telephony and Internet
functions Supports multiple call legs (i.e., forking)
In order to understanding SIP protocol, we need
bearing in mind that
 VoIP = Signaling +Media
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Protocol Stack
SIP
G.711,G.729
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SIP Architecture
SIP Request
SIP Response
RTP Media Stream
Redirect Server
Proxy Server
Location Server
Proxy Server
Proxy Server
User Agent Client(Caller)
User Agent Server(Callee)
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Components of SIP
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User Agents:
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User Agent Client (UAC) – Send Request (eg. invite…etc)
User Agent Server (UAS) - Responds to clients’ requests (eg.
successful ..etc)
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Server
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Proxy Server
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Registrar Server
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Receive SIP registration request
update user agent’s information
Redirect Server
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Receive SIP request from UA or other proxy
Forward or proxies the request to another location
Receive request from UA or proxy and returns a redirection
response(3xx)
Location Server
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It contain user URL,IP address script, feature
Routing information about Proxies,gateways and other Location
server in SIP network
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SIP basic operation modes
Proxy Mode
Proxy Server
Peer to Peer Mode
RTP
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SIP Operation in User Registration
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SIP Operation in Proxy Mode
Router
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SIP Operation in Redirect Mode
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SIP message structure
Request = Request-Line *
( general-header | request-header | entity-header )
CRLF
[message-body]
Request-Line = Method SP Request-URI SP SIP-Version CRLF
Response = Status-Line *
( general-header | response-header | entity-header )
CRLF
[ message-body ]
Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
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Request-Line of request message
Request-Line = Method SP Request-URI SP SIP-Version CRLF
Method :
 Register : Register to Registrar
 Invite
: Invite someone to participate in a session
 Cancel : Cancel the invitation
 Bye
: Finish the call
 ACK
: Request confirm
 Options : Query a server to its capabilities
Request-URI = SIP-URL without parameter or header element
SIP-Version = ”SIP/2.0”
EX :
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Status Line of Response message
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Status-Line = SIP-version SP Status-Code SP Reason-Phrase
CRLF
SIP-Version = ”SIP/2.0”
Status-Code :
 provisional response
1xx – Informational
 final response
2xx - Successful
3xx - Redirection
4xx - Request Failure
5xx - Server Failure
6xx - Global Failures
Reason-Phrase = “Trying”, “Ringing”…
Ex : SIP/2.0 100 Trying
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Complete list of response codes
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Request message Headers
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general-header
Apply to both request and response
messages
-Ex: Via, From, To, Call-ID, CSeq,
Contact, User-Agent …
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request-header
The additional information about the
request
-Ex: Proxy-Authorization, MaxForwards, …
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entity-header
Define meta-info about the message body
-Ex: Content-Length, Content-Type
The resource identified by the request
-Ex: Allow, Expires
*Refer to RFC 2543 Section 4.1
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SIP Header field
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Message Body
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v=0
o=UserB 2890844527
2890844527 IN IP4 there.com
s=Session SDP
c=IN IP4 110.111.112.113
t=0 0
m=audio 3456 RTP/AVP 0
a=rtpmap: 0 PCMU/8000
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SDP Specification
Session description
v= (protocol version)
o= (owner/creator and session identifier).
s= (session name)
i=* (session information)
u=* (URI of description)
e=* (email address)
p=* (phone number)
c=* (connection information - not required if included in all media)
b=* (bandwidth information)
One or more time descriptions (see below)
z=* (time zone adjustments)
k=* (encryption key)
a=* (zero or more session attribute lines)
Zero or more media descriptions (see below)
Time description
t= (time the session is active)
r=* (zero or more repeat times)
Media description
m= (media name and transport address)
i=* (media title)
c=* (connection information - optional if included at session-level)
b=* (bandwidth information)
k=* (encryption key)
a=* (zero or more media attribute lines)
*Refer to RFC 2327 Section 6
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SIP extensions list
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PSTN and SIP
interworking
Architecture of IP network
communication with PSTN
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Voice IP Protocol overview
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Interworking between PSTN and IP
network ISUP call control signaling
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SIP-T
What is SIP-T?
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SIP-T(SIP for Telephones) is define by IETF
Not an extension to SIP – a set of practices for
interfacing SIP to the PSTN, It provides two key
characteristics
 Encapsulation of ISUP in SIP
 Translation of ISUP parameters
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to SIP headers
Implemented at PSTN gateways, and carried
end-to-end
SIP-T Specification families :
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RFC3372 : SIP for Telephones (SIP-T): Context and
Architectures
RFC3398 : ISUP to SIP Mapping
RFC2976 : The Session Initiation Protocol (SIP) INFO Method
RFC3204 : MIME media types for ISUP and QSIG Objects
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IP network interworking with PSTN
using SIP-T architecture
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Translation of SIP-T
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SIP-T scenario 1
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SIP-T scenario 2
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Encapsulation of ISUP in SIP-T
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Extending SIP-T by encoding SS7 ISUP signaling messages allows MGCs
using SIP-T to be compatible with the PSTN.
SIP-T encodes and transmits the native signaling messages from one SCN
to another. To do this, SIP-T has been extended with MIME encoding of
signaling messages. The PSTN signaling messages are appended to the
SIP-T messages (such as INVITE, ACK, BYE) using binary encoding.
The use of MIME encoding with content type: APPLICATION allows PSTN
signaling messages to be tunneled between MGCs. The use of content
SUBTYPE enables SS7 ISUP messages to be differentiated by the
receiving MGC.
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SIP-T scenario 3
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PSTN-SIP v.s PSTN-SIP-PSTN
INVITE sip:[email protected]/user=phone SIP/2.0
Via : SIP/2/0/UDP gw1/carriwe.com:5060
From : sip:[email protected],.com;user=phone
To: sip:[email protected]/user=phone
Call-ID:[email protected]
CSeq: 1 INVITE
Contact : sip:[email protected],.com;user=phone
Content-Type: application/sdp
Content-length: 156
v=0
o= GATEway1 2890844527 2890844527 IN IP4 gatewayone.carrier.com
s=Session SDP
c= IN IP4 gatewayone carrier.com
t= 0 0
m= audio 3456 RTP/AVP 0
a = rtpmap:0 PCMU/8000
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SIP-I
Specification of SIP-I
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SIP-I (SIPwithEncapsulatedISUP ) is defined
by ITU-T (only draft until now)
ITU-T Series Q Supplement 45: Technical
Report TRQ.2815(Requirements for Interworking BICC/ISUP
Network with Originating/Destination Networks based on Session
Initiation Protocol and Session Description Protocol)
 定義了SIP與BICC/ISUP
互通時的技術需求，包括閘道
器類型、介接單元(Interworking Unit) 所應支援的協
定能力配置集與閘道器的安全模型等。
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ITU-T Recommendation Q.1912.5(Interworking
between Session Initiation Protocol (SIP) and the Bearer Independent Call
Control Protocol or ISDN User Part)
 Q.1912.5則定義
介接程序。
SIP與BICC/ISUP在介接單元的信號
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TRQ. 2815
Defines the signaling interworking between the Bearer Independent
Call Control (BICC) or ISDN User Part (ISUP) protocols and Session
Initiation Protocol (SIP) with its associated Session Description
Protocol (SDP) at an Interworking Unit (IWU)
Profile A was
defined to satisfy
the demand
represented by
3GPP in TA
24.229 V5.1.0
Profile C supports the trunking of
traffic via transit SIP networks using
MIME encoded encapsulatedISUP
(SIP-I)
Profile B complements Profile A, and both
of them are intended to support traffic
that terminates within the SIP network.
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Scenario
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Q.1912.5
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ISUP to SIP/SDP mapping
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Message mapping
 Parameter mapping
 Scope of parameter
 Mapping of ISUP parameters to SIP/SDP
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Initial address message (IAM) mapping to SIP
Encapsulation
SIP Network
ISUP
O-IWU
SIP-I
Scenario
I-IWU
ISUP
SIP header
ISUP message encapsulated in ful
Encapsulation format
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ISUP- SIP-I Message mapping
ISUP Acro
ISUP Message name
SIP message
GRA
Circuit group reset acknowledgement
GRS
Circuit group reset
BYE 500 Server Internal Error
IAM
Initial address
INVITE
IDR
Identification request
INFO 183 Session progress
IRS
Identification response
INFO 183 Session progress
INF
Information
INR
Information request
LPA
Loop back acknowledgement
LOP
Loop prevention
INFO 183 Session progress
NRM
Network resource management
INFO 183 Session progress
OLM
Overload
PAM
Pass-along
PRI
Pre-release information
INFO 183 Session progress
REL
Release
BYE Message codes]
RES
Resume
INFO 183 Session progress
RLC
Release complete
BYE (note)
BYE 500 Server Internal Error
ACM
Address complete
180 Ringing 183 Session progress
(profile C only)
ANM
Answer
200 OK INVITE
APM
Application transport
INFO or 183 Session progress
BLA
Blocking acknowledgement
ISUP side only
BLO
Blocking
ISUP side only
CCR
Continuity check request
ISUP side only
CFN
Confusion
INFO or 183 Session progress
CGB
Circuit group blocking
BYE 500 Server Internal Error
CGBA
Circuit group blocking ACK
ISUP side only
CGU
Circuit group unblocking
ISUP side only
CGUA
Circuit group unblocking ACK
ISUP side only
CON
Connect
200 OK INVITE
COT
Continuity
UPDATE
CPG
Call progress
180 Ringing 183 Session progress
(profile C only)
CRG
Charge information
RSC
Reset circuit
CQM
Circuit group query
SAM
Subsequent address
CQR
Circuit group query response
SDM
Subsequent directory number
DRS
Delayed release (reserved –
used in 1988 version)
SGM
Segmentation
Reassembled message ncapsulated.
SUS
Suspend
INFO 183 Session progress
FAA
Facility accepted
INFO 183 Session progress
UBL
Unblocking
FAC
Facility
INFO 183 Session progress
UBA
Unblocking acknowledgement
FAR
Facility request
INFO 183 Session progress
UCIC
Unequipped circuit identification code
FOT
Forward transfer
INFO 183 Session progress
UPA
User part available
FRJ
Facility reject
INFO 183 Session progress
UPT
User part test
ISUP side only
USR
User-to-user information
INFO 183 Session progress
NOTE: The Release Complete message is a link specific message and may invoke
BYE on some legs.
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Summary
Protocol summary
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SIP interworking capability “profiles”
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Conclusion
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軟交換與終端之間的控製協議方面，SIP是趨勢；
軟交換與應用服務器之間，SIP是主流;
SIP-I協議族的內容遠遠比SIP-T的內容要豐富。SIP-I協議族
不僅包括了基本呼叫的互通，還包括了CLIP、CLIR等補充
業務的互通；除了呼叫信令的互通外，還考慮到了資源預
留、媒體訊息的轉換等；既有固網軟交換環境下SIP與
BICC/ISUP的互通，也有移動3GPPSIP與BICC/ISUP的互通，
等等。
SIP-I協議族具有ITU-T標準固有的清晰準確和詳細具體，
可操作性非常強。
並且3GPP已經採用Q.1912.5作為IMS與PSTN/PLMN互通的
最終標準。所以，軟交換SIP域與PSTN的互通應該遵循
ITU-T的SIP-I協議族。實際上已經有許多電信運營商最終
選擇了SIP-I而放棄了SIP-T。
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Reference
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Keith Mainwaring :IP Telephony Migration Challenges
PCC.I-TEL/doc.0202/03, “Next Generation Networks Standards Overview (September 2003)”
ITUT Signalling Protocols for NGN Signalling Protocols
for NGN
RFC3398-ISUP to SIP Mapping
RFC3372-SIP-T-Context and Architectures
ITU-T Recommendation Q.1912.5, modified
王培元 :新世代電信網路電信級分封電話信號系統之效能
模式及分析
中國IT 實驗室 :软交换协议比较和发展趋势
Henry Sinnreich : Delivering VoIP and Multimedia
Service with Session Initiation Protocol
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