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7/24/2019 Sip session intial protocol
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Session Initiation Protocol
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Goal We want to know
SIP Architecture SIP Messages Syntax
Basic Call Example Session Description Protocol (SDP)
SIP Extensions
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TCP
IPv4/IPv6 support mobility
UDP
Q
oS
RTSPH.323......Web
Applications
RTCP RTPSDP
SIP Megaco
Wireless Access
3GPP/3GPP2iMAX W i Fi
Layer 2
Layer 1
Media
Audio, Video
RSVP
ADSL Cable
Residential Access
Ethernet
Company Access
SignalingQuality of
ServiceMedia
Transport
Multimedia protocol stack
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SIP Architecture
Introduction to SIP
SIP Network Entities
Simple Example
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Introduction
A powerful alternative to H.323 More flexible, simpler
Easier to implement Advanced features
Better suited to the support of intelligent userdevices
A part of IETF multimedia data and controlarchitecture
SDP (Session Description Protocol), RTSP (Real-Time Streaming Protocol), SAP (SessionAnnouncement Protocol)
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The Popularity of SIP
Originally Developed in the MMUSIC (MultipartyMultimedia Session Control) A separate SIP working group
RFC 2543 The latest version: RFC 3261
SIP + MGCP/MEGACO The VoIP signaling in the future
bake-off or SIPit (SIP Interoperability Tests) Various vendors come together and test their products
against each other To ensure that they have implemented the specification
correctly
To ensure compatibility with other implementations
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SIP Basic (1/2)
SIP is a peer-to-peer protocol End-devices initiate sessions SIP is an application layersignaling protocol
Create, modify and terminate sessions
Applications can be voice, video, gaming, instantmessaging, presence, call control, etc.
SIP uses existing IETF protocols to provide: Message formatting (HTTP 1.1)
Name resolution and mobility (DHCP and DNS) Media (RTP) Media negotiation (Session Description Protocol
SDP) Application encoding (MIME)
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SIP Basic (2/2)
SIP is ASCII text-based Eases implementation and debugging Uses URI style addresses and syntax Flexible transport
can use UDP, TCP, TLS, or SCTP Uses SDP for describing media sessions
Audio, video, realtime text ...
Simple extensible protocol
MethodsDefine transaction HeadersDescribe transaction
BodySDP and other MIME content
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SIP Architecture
A signaling protocol The setup, modification, and tear-down of multimedia
sessions
SIP + SDP
Describe the session characteristics Separate signaling and (RTP) media streams
SIP Signaling
IP Network
RTP Media Stream
SIP User SIP User
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SIP Network Entities
Client-Server Model Clients
User agent clients (UAC)
Application programs sending SIP requests Servers
Responds to clients requests
Four types of servers
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Types of Servers (1/3)
Proxy serversProxy servers Handle requests or forward requests to other servers
Can be used for call forwarding, time-of-day routing, orfollow-me services
[email protected] [email protected]
1.Request [email protected]
2.Request [email protected]
4.Response 3.Response
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Types of Servers (2/3)
Redirect serversRedirect servers Map the destination address to zero or more new
addresses
Redirect Server
1.Request [email protected]
2.Moved temporarily Contact: [email protected]
3.ACK
4.Request
5.Response
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Types of Servers (3/3)
A user agent server (UAS)A user agent server (UAS) Accepts SIP requests and contacts the user
The user responds an SIP response
Usually, a SIP device = UAC+UAS
A registrarA registrar Accepts SIP REGISTER requests
Indicating that the user is at a particular address
Personal mobility
Typically combined with a proxy or redirect server
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SIP Call Establishment
It is simple, which contains a number of interimresponses.
SIPdevice A
SIPdevice B
a
b
c
d
e
f
g
INVITE
Ringing
OK
ACK
Conversation
BYE
OK
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Call Completion to Busy SubscriberService
a
b
c
d
e
f
g
Conversation
BYE
OK
ACK
OK
Ringing
INVITE
Busy (Try at 4pm)
INVITE
ACK
j
i
h
SIP
device A
SIP
device B
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SIP Advantages
Attempt to keep the signaling as simple as possible Offer a great deal of flexibility
Does not care what type of media is to be exchangedduring a session or the type of transport to be used for the
media Various pieces of information can be included within
the messages Including non-standard information
Enable the users to make intelligent decisions
The control of the intelligent features is placed in the handsof the customer, not the network operator.
E.g., SUBJECT header
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Overview of SIP Messaging Syntax
SIP Message Syntax
SIP Request Messages
Simple Example
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SIP Messaging Syntax (1/2)
Text-based Similar to HTTP (Hypertext Transfer Protocol)
Disadvantage more bandwidth consumption
SIP messages message = start-line
*message-header CRLF
[message-body]
start-line = request-line | status-line Request-line specifies the type of request
Response line (status-line) indicates the success orfailure of a given request.
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SIP Messaging Syntax (2/2)
Message headers Additional information of the request or response
E.g.,
From: /To: headers for the originator and recipient
Retry-after header Subject header
Message body Describe the type of session
The most common structure for the message body is SDP(Session Description Protocol).
Could include an ISDN User Part message
Examined only at the two ends
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SIP Requests
Method SP Request-URI SP SIP-version CRLF
Ex: INVITE sip:[email protected] SIP/2.0
Request-URI
The address of the destination
Methods RFC 2543: INVITE,ACK, OPTIONS, BYE, CANCLE, REGISTER
RFC 2976: INFO
RFC 3261: SUPPORT,
RFC 3262: PRACK RFC 3265: SUSCRIBE & NOTIFY
RFC 3311: UPDATE
RFC 3428: MESSAGE
RFC 3515: REFER
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SIP Methods (1/2)
INVITEINVITE Initiate a session
Information of the calling and called parties
The type of media
Similar to IAM (initial address message) of ISUP
ACKACK Only when receiving the final response
BYEBYE Terminate a session
Can be issued by either the calling or called party
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SIP Methods (2/2)
OPTIONSOPTIONS Query a server as to its capabilities
A particular type of media
CANCELCANCEL Terminate a pending request E.g., an INVITE did not receive a final response
REGISTERREGISTER
Log in and register the address with a SIP server all SIP servers multicast address (224.0.1.175)
Can register with multiple servers
Can have several registrations with one server
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One number ServiceRegistrar/Proxy CallerUser at Address 1User at Address 2
OK
Register (address 1)
Register (address 2)
OK
INVITE
Trying
INVITE
INVITE
OK
CANCEL
OK (for INVITE)
OK (for CANCEL)
ACK
ACK
Conversation
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SIP Responses
SIP-Version SP Status-Code SP Reason-PhraseCRLF
Request: INVITE sip:[email protected] SIP/2.0
Response: SIP/2.0 200 ok
Status-Code A three-digit number
Reason-Phrase A textual description of the outcome
Could be presented to the user
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Status Code
Provisional response 1XX Informational (ex: 181 Ringing)
Final response 2XX Success (ex: 200 OK)
3XX Redirection (ex: 302 Moved temporarily)
4XX Client Error (ex: 401 Unauthorized)
5XX Server Error (ex: 505 SIP version not supported)
6XX Global Failure (ex: 604 Does not exist anywhere)
All responses, except for 1XX, are considered finaland Should be ACKed.
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SIP Addressing
SIP URIs (Uniform Resource Identifier)
user@host
Also called as URLs (Uniform Resource Locators)
Two types of SIP URIs:
Address of Record (AOR) (identifies a user) sip:[email protected]
Contact (identifies a device and is usually a Fully QualifiedDomain Name, FQDN) sip:[email protected] or sip:[email protected]
Other example:
sip:[email protected];user=phone
sips:[email protected]
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Message HeadersMessage Headers
Provide further information about the message Ex: Subject: vacation
Ex: Content-Type:application/sdp
Four main categories General, Request, Response, and Entity headers
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General Headers
Used in both requests and responses
To, From, CSeq, Call-ID, and Via are mandatory for all SIPmessages.
Basic information
To: headerin a REGISTER indicates the address-of-record ofthe user.
Ex: To: [email protected]
To: headerin an INVITE indicates the called party.
Ex: To: Boss
Ex: To: Boss
From: headerindicates the originator.
Ex: From: Daniel;tag=4455
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To: Header
The To headerfield first and foremost specifies
The desired "logical" recipient of the request
The address-of-record of the user or resource that is the target ofthis request.
The original recipient may or may not be the UAS processingthe request, due to call forwarding or other proxy operations.
Request-URI identifies the UAS that is to process the request.
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Request-URI
Request-URI identifies the UAS that is to processthe request.
The initial Request-URI of the message SHOULDbe set to the same value of the URI in the To field.
One notable exception is the REGISTER
A UA uses the REGISTER method to bind itsaddress-of-record to a specific contact address.
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Contact: Header
Provides a URI for use in future communication regarding aparticularsession
Ex1: In a SIP INVITE, the Contact header might be differentfrom the From header.
An third-party administrator initiates a multiparty session.
Ex 2: Used in response, it is useful for directing furtherrequests directly to the called user.
Ex 3: It is used to indicate a more appropriate address if anINVITE issued to a given URI failed to reach the user.
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Request / Response Headers
Request Headers
Apply only to SIP requests
Addition information about the request or the client
Ex: Subject: vacation
Ex: Priority: (emergency, urgent, normal, or non-urgent)
Response Headers
Further information about the response that cannot be included
in the status line Ex: Unsupported
Ex: Retry-After
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Entity Headers
Indicate the type and format of information includedin the message body
Content-Length: the length of the message body
Content-Type: media type of the message body Ex: Content-Type:application/sdp
Content-Encoding: for message compression
Content Disposition: how a message part should beinterpreted
session, icon, alert, render
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Examples
Registration
Invitation
Termination of a Call
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REGISTER sip:registrar.work.com SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123
Max-Forwards: 70From: sip:[email protected]; tag = 123456To: sip:[email protected]: 123456@ station1.work.comCSeq: 1 REGISTERContact: sip:[email protected]: 7200Content-Length: 0
SIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: sip:[email protected]; tag = 123456To: sip:[email protected]: 123456@ station1.work.comCSeq: 1 REGISTERContact: sip:[email protected]: 3600Content-Length: 0
a
b
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Invitation
A two-party call
Subject:
optional
Content-Type:
application/sdp
Transaction: Command Sequence
A dialog ID
To identify a peer-to-peer
relationship between twouser agents
Tag in From
Tag in To
Call-ID
BossDaniel
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551Contact: sip:[email protected]: BossCall-ID: [email protected]: 1 INVITESubject: VacationContent-Length: xxxContent-Type: application/sdp
Content-Disposition: session(message body)
a
bSIP/2.0 180 RingingVia:SIP/2.0/UDP station1.work.com;branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Contact: sip:[email protected]: [email protected]: 1 INVITEContent-Length: 0
cSIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Contact: sip:[email protected]: [email protected]: 1 INVITESubject: VacationContent-Length: xxxContent-Type: application/sdpContent-Disposition: session(message body)
dACK sip:[email protected] SIP/2.0Via:SIP/2.0/UDP station1.work.com;branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 1 ACKContent-Length: 0
Conversation
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Termination of a CallBossDaniel
BYE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 2 BYEContent-Length: 0
a
bSIP/2.0 200 OK
Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 2 BYEContent-Length: 0
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Redirect and Proxy Servers
Redirect serverProxy server
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Redirect server
An alternative address 302 Moved temporarily
Another INVITE
Same Call-ID CSeq ++
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK124Max-Forwards: 70From: Daniel; tag=44551Contact: sip:[email protected]: BossCall-ID: [email protected]: 2 INVITESubject: Vacation
Content-Length: xxxContent-Type: application/sdpContent-Disposition: session(message body)
c
d
ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551To: BossCall-ID: [email protected]: 1 ACK
Boss
Daniel
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: Daniel; tag=44551Contact: sip:[email protected]
To: BossCall-ID: [email protected]: 1 INVITESubject: VacationContent-Length: xxxContent-Type: application/sdpContent-Disposition: session(message body)
a
bSIP/2.0 302 Moved Temporarily
Via:SIP/2.0/UDP station1.work.com;branch=z9hG4bK123From: Daniel; tag=44551To: Boss; tag=11222Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]
Redirect Server
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Proxy server
Sits between a user-agent client and the far-enduser-agent server
Numerous proxies can reside in a chain betweenthe caller and callee.
The proxy may change the Request-URI. Via: header
The path taken by a request
Loop detected, 482 (status code) For a response
The 1st Via: header, Checked, Removed
Branch: used to distinguish between multiple responses to
the same request/ to detect loop
p4p3p2p1caller
INVITE
482
ACK
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Forking server
Forking Proxy: Issue a single request to multipledestinations A user is registered at several locations
fork requests
Branch: used to distinguish between multipleresponses to the same request Ex: on Via: header ;branch=z9hG4bK123
In order to handle such forking, a proxy must bestateful.
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Proxy state
Transaction stateless Proxy server forwards all methods and responses without
interaction
Transaction stateful Maintains state for the transaction until the final response
is received
Transaction Control Block (TCBTCB)
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Record-Route: and Route:
Record-Route:
Use Contact: the messages may not pass through thesame proxy
A stateful proxy might require that it remains in thesignaling path
Insert its address into the Record-Route: header
The response includes the Record-Route: header
The information contained in the Record-Route: header isused in the subsequent requests related to the same call.
The Route: header = the Record-Route: header inreverse order
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Through a Proxy (1/2)Boss Daniel
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Max-Forwards: 70From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]
CSeq: 1 INVITE
a
b
SIP/2.0 100 TryingVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890From: Boss; tag=ab12To: DanielCall-ID: [email protected]: 1 INVITE
c
sip:Server.work.com
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890
Max-Forwards: 69Record-route: From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]: 1 INVITE
SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Record-route: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]
SIP/2.0 200 OK...
e
d
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Through a Proxy (2/2) Boss Daniel
ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7891Max-Forwards: 70Route: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]
CSeq: 1 ACK
e
f
sip:Server.work.com
ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz2Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7891Max-Forwards: 69From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 ACK
SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Record-route: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]
CSeq: 1 INVITEContact: sip:[email protected]
g
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Multiple Locations (1/2)Boss
pc1
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Max-Forwards: 70From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]: 1 INVITE
a
b
SIP/2.0 100 TryingVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789From: Boss; tag=ab12To: DanielCall-ID: [email protected]: 1 INVITE
c
sip:Server.work.com
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bK123Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Max-Forwards: 69Record-coute: From: Boss; tag=ab12Contact: BossTo: Daniel
Call-ID: [email protected]: 1 INVITEd
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bK456Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Max-Forwards: 69Record-coute: From: Boss; tag=ab12Contact: BossTo: DanielCall-ID: [email protected]: 1 INVITE
pc2
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Multiple Locations (2/2) Boss
pc1
e
f
g
sip:Server.work.com
CANCEL sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bK456Max-Forwards: 69Record-coute: From: Boss; tag=ab12
Contact: BossTo: DanielCall-ID: [email protected]: 1 CANCEL
SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bK456Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Record-coute: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]
pc2
SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK789Record-coute: From: Boss; tag=ab12To: Daniel; tag=xyz45Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]
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Session Description Protocol (SDP)
Introduction to SIP
SIP Network Entities
Simple Example
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The Session Description Protocol
The Most Common Message Body Be session information describing the media to be
exchanged between the parties
SDP, RFC 2327 (initial publication) (3266 4566)
SDP simply provides a format for describing sessioninformation to potential session participants.
RFC 3264 (An Offer/Answer Model with SDP) SIP uses SDP in an answer/offer mode.
An agreement between the two parties as to the types ofmedia they are willing to share
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The Structure of SDP
Text-based Protocol
The Structure of SDP Session Level Information
Name of the session
Originator of the session Time that the session is to be
active
Media Level Information
Media type Port number
Transport protocol
Media format
Originator and Session ID
Protocol Version
Session Name
Session Time
Media Name and Transport
Connection Information
Media Name and Transport
Connection Information
Session Description
Session Level Information
Media Description 1
Media Description 2
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SDP Syntax
A number of lines of text
In each line field=value
Ex: v=0
field= ...
Ex: m=audio 45678 RTP/AVP 0
field is exactly one character(case-significant)
Session-level fields Begin with version description field (v=)
Media-level fields
Begin with media description field (m=)
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Mandatory Fields (1/3)
v, o, s, t, m are mandatory fields.
v=(protocol version) (ex: v=0)
s=(session name) (ex: s= )
A text string for multicast conference
t=(time of the session) (ex: t=0 0)
The start time and stop time for pre-arranged multicast
conference
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Mandatory Fields (2/3) o=(session origin or creator)
Ex: o=collins 123456 001 IN IP4 s1.yy.com Username (ex: collins): the originators login id or -
Session ID (ex: 123456)
Make use of NTP timestamp as a unique ID
Version (ex: 001) Network type (ex: IN)
IN refers to Internet
Address type (ex: IP4)
IP4, IP6
address (ex: s1.yy.com)
A fully-qualified domain name or the IP address
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Mandatory Fields (3/3)
m=(media) Ex: m=audio 45678 RTP/AVP 15 3 0
Media type (ex: audio) Audio, video, application, data, control
The transport port (ex: 45678)
The transport protocol (ex: RTP/AVP)
The media format, an RTP payload format
(ex: 15 3 0)
List the various types of media format that can be supported
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Optional Fields (1/4)
Some optional fields can be applied at both session and
media levels. The value applied at the media level overrides that at the
session level for that media instance.
c=(connection data) (ex: c=IN IP4 s1.yy.com)
The network and address at which media data will be received
Network type (ex: IN)
Address type (ex: IP4)
Connection address (ex: s1.yy.com)
At session or media level
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Optional Fields (2/4)
a=(attributes) (ex: a=send only) Describe additional attributes
Property attribute
Ex: a=sendonly / a=recvonly
Value attribute
Used in a shared whiteboard session
Ex: a=orient:portrait / a=orient:landscape
rtpmap provides additional information for dynamic payloadtype.
Ex: m=video 54678 RTP/AVP 98
a=rtpmap:98 L16/16000/2
Dynamic RTP payload type=98
16-bit linear encoded stereo (2 channels) audio sampled at 16kHz
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Optional Fields (2/4)
i=(session information)
A text description At both session and media levels
It would be somewhat superfluous, since SIP already supports theSubject header.
u=(URI of description) Where further session information can be obtained
Only at session level
e=(e-mail address)
Who is responsible for the session Only at the session level
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Optional Fields (3/4)
p=(phone number)
Only at the session level
b=(bandwidth information)
In kilobits per second
At session or media level
r=(repeat times)
For regularly scheduled session a session is to be repeated
How often and how many times
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Optional Fields (4/4)
z=(timezone adjustments)
For regularly scheduled session Standard time and daylight savings time
k=(encryption key)
An encryption key or a mechanism to obtain it for the purposesof encrypting and decrypting the media
At session or media level
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Ordering of Fields Session Level
Protocol version (v)
Origin (o)
Session name (s)
Session information (i)
URI (u)
E-mail address (e) Phone number (p)
Connection info (c)
Bandwidth info (b)
Time description (t)
Repeat info (r)
Time zone adjustments (z)
Encryption key (k)
Attributes (a)
Media level
Media description (m)
Media info (i)
Connection info (c)
Optional if specified at thesession level
Bandwidth info (b)
Encryption key (k)
Attributes (a)
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Usage of SDP with SIP
Simple Example
Negotiation of Media
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Usage of SDP with SIP
SIP and SDP make a wonderful partnership for the
transmission of session information. SIP provides the messaging mechanism for the establishment
of multimedia sessions.
SDP provides a structured language for describing thesessions.
The entity headers identifies the message body.
Ex: Content-Type:application/sdp
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Offer/Answer Model
Offer a selection of media formats
Answer which the receiver is willing to accept Supported media type: returned with a transport port
Unsupported media type: also be returned with a port
number of zero
(1) Offer
(2) Answer
Sender Receiver
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Daniel Boss
INVITE sip:[email protected] SIP/2.0From: Daniel; tag = abcd1234To: BossCSeq: 1 INVITEContent-Length: 213
Content-Type: application/sdpContent-Disposition: session
v=0o=collins 123456 001 IN IP4 station1.work.coms=c=IN IP4 station1.work.com
t=0 0m=audio 4444 RTP/AVP 2a=rtpmap 2 G726-32/8000m=audio 4666 RTP/AVP 4a=rtpmap 4 G723/8000m=audio 4888 RTP/AVP 15a=rtpmap 15 G728/8000
a
b
SIP/2.0 200 OK
SDP Inclusion in SIP Messages (1/2)
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SIP/2.0 200 OKFrom: Daniel; tag = abcd1234To: Boss; tag = xyz789CSeq: 1 INVITEContent-Length: 163Content-Type: application/sdp
Content-Disposition: session
v=0o=collins 45678 001 IN IP4 station2.work.coms=c=IN IP4 station2.work.comt=0 0
m=audio 0 RTP/AVP 2m=audio 0 RTP/AVP 4m=audio 6666 RTP/AVP 15a=rtpmap 15 G728/8000
b
c
d
ACK sip:[email protected] SIP/2.0From: Daniel; tag = abcd1234To: Boss; tag = xyz789CSeq: 1 ACK
Content-Length: 0
Conversation
Daniel Boss
SDP Inclusion in SIP Messages (2/2)
port=4888 port=6666
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Alternative Ways
Offerm=audio 4444 RTP/AVP 2 4 15a=rtpmap 2 G726-32/8000
a=rtpmap 4 G723/8000a=rtpmap 15 G728/8000
Answerm=audio 6666 RTP/AVP 15a=rtpmap 15 G728/8000
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Negotiation of Media (1/3)
Re-INVITE is issued when the server replies with
more than one codec. With the same dialog identifier(To and From tag), Call-ID
and Request-URI
Session version in o field is increased by 1.
A mismatch 488 (Not acceptable here) or 606 (Not acceptable)
A Warning header with warning code 304 (media type not available)
305 (incompatible media format)
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Negotiation of Media (2/3)INVITE sip:[email protected] SIP/2.0CSeq: 1 INVITEContent-Length: 183
Content-Type: application/sdpContent-Disposition: session
v=0o=collins 123456 001 IN IP4 station1.work.coms=c=IN IP4 station1.work.comt=0 0m=audio 4444 RTP/AVP 2 4 15a=rtpmap 2 G726-32/8000a=rtpmap 4 G723/8000a=rtpmap 15 G728/8000a=inactive
Daniel Boss
b
a
SIP/2.0 200 OK
CSeq: 1 INVITE
Content-Length: 157
Content-Type: application/sdp
Content-Disposition: session
v=0
o=coll ins 45678 001 IN IP4 station2.work.com
s=
c=IN IP4 station2.work .com
t=0 0
m=audio 6666 RTP/AVP 4 15
a=rtpmap 4 G723/8000
a=rtpmap 15 G728/8000a=inactive
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Daniel Boss
d
c
INVITE sip:[email protected] SIP/2.0CSeq: 2 INVITEContent-Length: 126Content-Type: application/sdpContent-Disposition: session
v=0o=collins 123456 002 IN IP4 station1.work.coms=c=IN IP4 station1.work.comt=0 0m=audio 4444 RTP/AVP 15
a=rtpmap 15 G728/8000
ACK sip:[email protected] SIP/2.0From: Daniel; tag = abcd1234To: Boss; tag = xyz789CSeq: 1 ACKContent-Length: 0
Negotiation of Media (3/3)
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Determine the capabilities of a potential called
party Accept Header
Indicate the type of information that the sender hopes toreceive
Allow Header Indicate the SIP methods that it can handle
Supported Header Indicate the SIP extensions that it can be supported
OPTIONS Method
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Daniel Boss
b
aOPTIONS sip:[email protected] SIP/2.0Via: SIP/2.0/UDP Station1.work.com; branch=z9hG4bK7890123From: Daniel; tag=lmnop123To: BossCall-ID: [email protected]: DanielCSeq: 1 OPTIONSAccept: application/sdpContent-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP Station1.work.com; branch=z9hG4bK7890123
From: Daniel; tag=lmnop123
To: Boss; tag=xyz5678
Call-ID: [email protected]
CSeq: 1 OPTIONSAl low: INVITE, ACK, CANCEL, OPTIONS, BYE
Supported: newfield
Content-Length: 146
Content-Type: application/sdp
v=0
o=manager 45678 001 IN IP4 station2.work.com
s=c=IN IP4 station2.work.com
t=0 0
Usage of the OPTIONS Method
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SIP Extensions and Enhancements
SIP INFO Method
SIP Event Notification
SIP for Instant Messaging
SIP REFERMethod
Reliability of Provisional Responses
SIP UPDATE Method
Integration of SIP Signaling and Resource Management
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SIP Extensions and Enhancements
RFC 2543, March 1999
SIP has attracted enormous interest.
Traditional telecommunications companies, cable TV providersand ISP
A large number of extensions to SIP have been proposed.
SIP will be enhanced considerably before it becomes an Internetstandard.
RFC 3261
183 Session Progress
Supported header
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183 Session Progress
It has been included within the RFC 3261.
To open one-way audio path from called end to calling end From the called party to calling party
Enable in-band call progress information to be transmitted
Tones or announcements
Interworking with SS7 network ACM (Address Complete Message)
Ring back tone in 183 Session Progress
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The Supported Header The Require Header (client server)
A client indicates that a server must support certain extension withRequire header
420 bad extension with Unsupported header
The Supported header
Way 1:
Options (client server)
Response with Supported (server client)
Way 2:
Request with Supported (client
server) 421 extension required with Require (server client)
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SIP INFO Method
Specified in RFC 2976
For transferring information during an ongoing session Application-layer information could be transferred in the middle
of a call.
A powerful, flexible tool to support new services
The transfer of DTMF digits
The transfer of account balance information
Pre-paid service
The transfer of mid-call signaling information generated inanother network and passed to the IP network via a gateway
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SIP Event Notification
RFC 3265 hasaddressed the issue ofevent notification.
SUBSCRIBE andNOTIFY
Event header
Subscriber Notifier
SUBSCRIBE
200 OK
NOTIFY
200 OK
NOTIFY
200 OK
a
b
c
d
e
f
Current stateinformation
Updated stateinformation
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SIP for Instant Messaging (1/3)
The IETF working group SIP for Instant Messaging
and Presence Leveraging Extensions (SIMPLE) RFC 3428 Session Initiation Protocol (SIP)
Extension for Instant Messaging
A new SIP method MESSAGE This request carries the actual message in a message body.
A MESSAGE request does not establish a SIP dialog.
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DanielBoss sip:Server.work.com
MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Max-Forwards: 70From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGEContent-Type: text/plainContent-Length: 19Content-Disposition: render
Hello. How are you?
MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890Max-Forwards: 69From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGEContent-Type: text/plainContent-Length: 19Content-Disposition: render
Hello. How are you?
SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bKxyz1Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGE
Content-Length: 0
SIP/2.0 200 OKVia: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890From: BossTo: DanielCall-ID: [email protected]: 1 MESSAGEContent-Length: 0
a
b
c
d
SIP for Instant Messaging (2/3)
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DanielBoss sip:Server.work.com
MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: DanielTo: BossCall-ID: [email protected]: 1101 MESSAGE
Content-Type: text/plainContent-Length: 22Content-Disposition: render
Im fine. How are you?
MESSAGE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKabcdVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 69From: DanielTo: Boss
Call-ID: [email protected]: 1101 MESSAGEContent-Type: text/plainContent-Length: 22Content-Disposition: render
Im fine. How are you?
SIP/2.0 200 OKVia: SIP/2.0/UDP server.work.com; branch=z9hG4bKabcdVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: DanielTo: BossCall-ID: [email protected]: 1101 MESSAGE
Content-Length: 0
SIP/2.0 200 OKVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123From: DanielTo: BossCall-ID: [email protected]
CSeq: 1101 MESSAGEContent-Length: 0
e
f
g
h
SIP for Instant Messaging (3/3)
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SIP REFER Method RFC 3515 The Session Initiation Protocol (SIP) Refer
Method
To enable the sender of the request to instruct the receiverto contact a third party With the contact details for the third party included within the REFER
request
For Call Transfer applications
The Refer-to: and Refer-by: Headers
The dialog between Mary and Joe remains established. Joe could return to the dialog after consultation with Susan.
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sip:[email protected] sip:[email protected] sip:[email protected]
REFER sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK789Max-Forwards: 70From: Mary; tag=123456To: Joe; tag=67890Contact: MaryRefer-To: Susan
Call-ID: [email protected]: 123 REFERContent-Length: 0
SIP/2.0 202 AcceptedVia: SIP/2.0/UDP station1.work.com; branch=z9hG4bK789From: Mary; tag=123456To: Joe; tag=67890
Contact: JoeCall-ID: [email protected]: 123 REFERContent-Length: 0
INVITE sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1Max-Forwards: 70
From: Joe; tag=abcxyzTo: SusanContact: JoeCall-ID: [email protected]: 567 INVITEContent-Type: application/sdpContent-Length: xx
Content-Disposition: session{message body}
a
b
c
Call Transfer (1/2)
Call Transfer (2/2)
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sip:[email protected]:[email protected]
fg
SIP/2.0 200 OKVia: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1From: Joe; tag=abcxyzTo: Susan; tag=123xyzCall-ID: [email protected]: 567 INVITE
Content-Type: application/sdpContent-Length: xxContent-Disposition: session{message body}
ACK sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1Max-Forwards: 70
From: Joe; tag=abcxyzTo: Susan; tag=123xyzCall-ID: [email protected]: 567 ACKContent-Length: 0
NOTIFY sip:[email protected] SIP/2.0Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bK123Max-Forwards: 70
To: Joe; tag=67890From: Mary; tag=123456Contact: JoeCall-ID: [email protected]: 124 NOTIFYContent-Type: message/sipfrag;version=2.0Content-Length: 15
SIP/2.0 200 OK
SIP/2.0 200 OKVia: SIP/2.0/UDP station2.work.com; branch=z9hG4bK123To: Joe; tag=67890From: Mary; tag=123456Call-ID: [email protected]
CSeq: 124 NOTIFYContent-Length: 0
h
( )
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Reliability of Provisional Responses (1/2)
Provisional Responses
100 (trying), 180 (ringing), 183 (session in progress) Are not answered with anACK
If the messages is sent over UDP
Unreliable
Lost provisional response may cause problems wheninteroperating with other network
180 (ring), 183 (session progress) Q931 alerting or ISUPACM (Address Complete Message)
These external networks need these information to drive a statemachine.
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Reliability of Provisional Responses (2/2)
E.g., a call to an unassigned number
ACM to create a one-way path to relay an announcement suchas The number you have called has been changed
If the provisional response is lost, the caller might left in the darkand not understand why the call did not connect.
RFC 3262 Reliability of Provisional Responses in SIP
Response sequence (RSeq): a request header
Response ACK (RAck): a response header
Option tag=100rel with supported:/unsupported:
Provisional Response ACK (PRACK) method
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Provisional Response ACK (1/2)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123
Supported: 100relRequire: 100rel
From: sip:[email protected]; tag=lmnop123
To: sip:[email protected]: [email protected]
CSeq: 1 INVITE
??SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123
Require: 100relRSeq: 567890
From: sip:[email protected]; tag=lmnop123
To: sip:[email protected]; tag = xyz123Call-ID: [email protected]
CSeq: 1 INVITE
Response
Lost
b
a
c
ResponseRetransmit
[email protected] [email protected]
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123
Require: 100rel
Start timer T1
default=0.5 sec
Provisional Response ACK (2/2)[email protected] [email protected]
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SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123Require: 100rel
RSeq: 567891
From: sip:[email protected]; tag=lmnop123
To: sip:[email protected]; tag = xyz123
Call-ID: [email protected]
CSeq: 1 INVITE
d
c
eSIP/2.0 200 OK
Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123
From: sip:[email protected]; tag=lmnop123
To: sip:[email protected]; tag=xyz123
Call-ID: [email protected]
CSeq: 2 PRACK
PRACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP ClientA.network.com; branch=z9hG4bK7890123
RAck: 567891 1 INVITE
From: sip:[email protected]; tag=lmnop123
To: sip:[email protected]; tag=xyz123
Call-ID: [email protected]
CSeq: 2 PRACK
Th SIP UPDATE M th d
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The SIP UPDATE Method
RFC 3311 The Session Initiation Protocol (SIP)
UPDATE Method To enable the modification of session information
before a final response to an INVITE is received
E.g., to change the codec One important usage is when reserving network
resources as part of a SIP session establishment
Integration of SIP Signaling and
R M t (1/2)
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Resource Management (1/2) Ensuring that sufficient resources are available to
handle a media stream is a very important.
The signaling might take a different path from themedia.
Assume resource-reservation mechanisms are
available End-to-end status
End-to-end network resources are reserved as part of sessionestablishment.
Segmented status A certain amount of network resources are reserved in
advance for a certain type of usage.
Policing functions at the edge of the network
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Usage of SIP for Features and Services
Call Forwarding
Consultation Hold
U f SIP f F t /S i (1/2)
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Usage of SIP for Features/Services (1/2) Personal Mobility (Registrar)
Call-transfer application (with REFER method)
One number service through forking proxy
Call-completion services (by Retry-After: header)
Click-to-call applications or Web service (SIP address
is a URL) Existing supplementary services in traditional
telephony Call waiting, call forwarding, multi-party calling, call screening
Usage of SIP for Feat res/Ser ices (2/2)
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Usage of SIP for Features/Services (2/2) Proxy invokes various types of advanced
feature logic. Policy server (call-routing, QoS)
Feature server (subscriber-specific feature data, ex:screening lists, forwarding information)
Authentication server Use the services of an IN SCP over INAP
Use Open Service Access (OSA)
Call Forwarding
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Call Forwarding Call-forwarding-
on-busy Call-forwarding-
on-no-answer Timeout
CANCEL method
Call-forwardingunconditional
INVITE sip:[email protected] SIP/2.0
From: sip:user1
To: sip:[email protected]
Contact: User1
CSeq: 1 INVITE
SIP/2.0 100 Trying
From: sip:user1
To: sip:[email protected]
CSeq: 1 INVITE
INVITE sip:[email protected] SIP/2.0
From: sip:user1
To: sip:[email protected]
Contact: User1
CSeq: 1 INVITE
SIP/2.0 486 Busy Here
From: sip:user1
To: sip:[email protected]
CSeq: 1 INVITE
INVITE sip:[email protected] SIP/2.0
From: sip:user1
To: sip:[email protected]
CSeq: 2 INVITE
SIP/2.0 200 OK
From: sip:user1
To: sip:[email protected]
Contact: sip:[email protected]
CSeq: 2 INVITE
SIP/2.0 200 OK
From: sip:user1
To: sip:[email protected]
Contact: sip:[email protected]
CSeq: 1 INVITE
User1 sip:Server.work.com
User2
User3
Consultation Hold INVITE sip:User A@there SIP/2.0
User A User B User C
a
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Consultation Hold
User A asks User B a
question, and User Bneed to check with UserC for the correct answer.
User B could use the
REFER method totransfer the call to UserC.
A SIP UPDATE
INVITE sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345SDP Description
SIP/2.0 200 OKFrom: sip:User B@hereCall-ID: 12345SDP Description
ACK sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345
Conversation
UPDATE sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345
Session Descriptiona=inactive
SIP/2.0 200 OKFrom: sip:User B@hereCall-ID: 12345Session Descriptiona=inactive
On hold
UPDATE sip:User A@there SIP/2.0From: sip:User B@hereCall-ID: 12345Session Descriptiona=sendrecv
SIP/2.0 200 OKFrom: sip:User B@hereCall-ID: 12345
Session Descriptiona=sendrecv
User B calls User C, speaks, then hangs up
b
c
d
e
f
gh
i
j
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Interworking
PSTN Interworking
SIP to PSTN [email protected] Proxy.work.com PSTN switchNGW
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INVITE
183 (Session Progress)Session description
One-way audio
a
b
g
i
j
m
100 (Trying)INVITE
100 (Trying)
183 (Session Progress)Session description
IAM
ACM
One-way audio
ANM
200 (OK)Updated session description
200 (OK)Updated session description
ACK
ACK
Two-way audio Two-way audio
c
de
f
h
k
l
n
o
IAM (initial address message)
ACM (Address Complete Message)
ANM (Answer Message)
PSTN Interworking
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PSTN Interworking
RFC 3372 - Session Initiation Protocol for
Telephones (SIP-T): Context and Architectures PSTN SIP PSTN
RFC 3204 - MIME media types for ISUP and QSIGObjects
MIME media types
For ISUP
The whole issue of interworking with SS7 is
fundamental to the success of VoIP in the realworld.
Summary
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Summary
The future for signaling in VoIP networks
Simple, yet flexible Easier to implement
Fit well with the media gateway control protocols
SIP is the protocol of choice for the evolution ofthird-generation wireless networks. SIP-based mobile devices will become available
SIP-based network elements will be introduced withinmobile networks.
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End.