Sde (5ess), Alttc: Nitin Garg
Sde (5ess), Alttc: Nitin Garg
NGN PROTOCOLS
NGN Protocols
SGW SS7 SS7 SIGTRAN IP Media Gateway Controller SIP/H.323 Media Gateway Controller Terminals
SS7
SGW
IP Media Gateway Controller Media Gateways MGCP/Megaco
PSTN
Terminals
What is SIP ?
What is PROTOCOL ?
A protocol is a set of rules to be used for communication between two entities. Elements of a protocol are : 1. Syntax: Refers to How part. The data format e.g.TCP/IP header 2. Semantics: Refers to What part. The control information communicated. 3. Timing: Refers to When part. Synchronous or asynchronous
What is a Session?
It is a temporary communication relationship among a group of objects in the service stratum that are assigned to collectively fulfill a task for a period of time A state of a session may change during its lifetime Session based communications may be one-to-one, one-to-many, many-to-one or many-to-many
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What is SIP ?
The Session Initiation Protocol (SIP) is an application layer Signaling Control Protocol used to establish, maintain and terminate multimedia sessions. Multimedia sessions include Internet Telephony, conferences and other similar applications involving such media as audio, video and data. SIP is a protocol from IETF and is defined in RFC 2543
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SIP Characteristics
User to User protocol Establishes, modifies, and tears-down sessions
Relies on other protocols for transport, QoS, Accounting Allows multi-party sessions; multi-casting
SIP Characteristics
SIP is dependent upon other protocols like RTP for media transport, Real Time Streaming Protocol (RTSP), Session Description Protocol (SDP) for negotiation of user capabilities and other parameters etc.
SIP Characteristics
SIP supports all facets of establishing and terminating multimedia communications:
User Location: determination of the end system to be used for communication User Capabilities: determination of the media and media parameters to be used User availability: determination of the willingness of the called party to engage in communication Call setup: ringing, establishment of call parameters at both called and calling party
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Types of Sessions
The term session is purposely ambiguous. SIP protocol does not care what type of session is being maintained. SIP sessions can be:
Telephone Calls Multimedia Conferences On line Transactions
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SIP Architecture
Redirect Server Proxy Server Terminals and gateways Are also Known as User Agents
PSTN
IP Network
Gateway
SIP Components
SIP basically has two components
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SIP Addressing
The SIP Uniform Resource Identifier (URI), sometimes called the Uniform Resource Locator (URL), identifies participants in a SIP session. SIP:user@host
Similar to HTTP URI Can designate an individual,
Sip:[email protected]
SIP:2145551212@gateway
Tel:214-555-1212
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SIP Messages
SIP messages are typically of type requests and responses. Requests flow from client to server and a response from server to a client SIP messages are send either in TCP or UDP. If the protocol type is not listed in URI the client must first attempt to connect using UDP and then TCP Being text based protocol, headers are largely self describing
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SIP Messages
The SIP request starts with a command line. The command line indicates the type of request that is being sent, a field request URI giving the detail of destination and the version of SIP protocol Several headers follow the command line
To and From (Calling and called party identity) Call-Id is a unique token that identifies session Via defines the path that the request has taken Cseq defines the sequence number of the request
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SIP Messages
The SIP message optionally contains a message body. The content and structure of message body is variable
The content type and content-length headers specify the form and size of the message body The message body will typically contain a Session Description Protocol (SDP) messages that contain information about the session, such as media coding format and IP address and port number
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Headers
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Headers
There are 44 SIP headers listed in RFC 2543. These headers can be divided into four different groups
General Headers Request Headers Response headers Entity Headers
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SIP Headers
General Headers
Call-ID, Contact, Cseq, Date, Encryption, From, RecordRoute, Timestamp, To, via
Request-Headers
Accept-Language, Authorization, Contact, Hide, MaxForward, Organization, Priority, Proxy-Authorization, Proxy-Require, Route, Require, Response-Key, Subject, User Agent
Response headers
Retry-After, Server, Unsupported, Warning, WWWAuthenticate
Entity-Header
Content-Type
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2xx Success
200 OK
3xx Redirect
301 moved permanently 302 moved temporarily
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Locating a server
The client must determine the IP address and port no of the server for which the request is destined If the port no is not listed in the SIP URL, the default port is 5060 If the protocol is not listed the client must first attempt using UDP and then TCP The client queries the Domain Name System (DNS) for the host IP address
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2 3
OK 200, Response
IP
SIP UA Client
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Conversation
SIP UA Server
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Proxy Server
Signaling
IP
User Agent Client User Agent Server
SIP Signaling
Media
NITIN GARG, SDE(5ESS), ALTTC
User Traffic
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Mumbai.tcss.co.in
Delhi.tcs.co.in
(2) vinod
vinodworkstation
(4) INVITE vinod@vinodworkstation
(6) 200 OK
(9) ACK vinod@vinodworkstation
(5)
Anilworkstation
PROXY Server
Vinod@vinodworkstation
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Switch PSTN
POTS
REL RLC 5 OK Response
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SIP Mobility
SIP provides full mobility to the user. User can move from one service provider to another service provider and can retain the same address. The caller may always use the same address, phone number or URL, but will be redirected transparently to the network, location or device of choice of the called party. User can move temporarily or permanently to another network.
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SIP Mobility
Mobility in an IP environment is classified as :
Personal mobility different terminals, same personal identity (address) Terminal mobility the ability to maintain communications when moving a single end system from one subnet to another Service mobility keep same services while mobile
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SIP Mobility
INVITE Request To: bob@usa Moved Temporarily Contact: bob@asia
2 REGIRSTR RequestTo:bob@usa Contact: bob@asia 1
IP
4
IP
ACK Request
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Proxy Server
IP
UAS
UAC
2. SIP/2.0 407 Proxy Authorization Reqd To: sip:bob@host From: ted@otherhost Call-id: 12345@otherhost Cseq: 1 INVITE Proxy-Authenticate.
Initiation Authorization
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Proxy Server
IP
UAS
UAC
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IP
UAS
UAC
4.c
SIP/2.0 180 Ringing To:sip:bob@host From: ted@otherhost Call-Id: 12345@otherhost Cseq: 1 INVITE
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5b. SIP/2.0 200 OK To: sip:bob@host From: ted@otherhost Call-id: 12345@otherhost Cseq: 1 INVITE
5c. ACK sip:bob@host SIP/2.0 To: sip:bob@host From: ted@otherhost Call-Id:12345@otherhost Cseq: 1 ACK
IP
UAS
UAC 5a. SIP/2.0 200 OK To:sip:bob@host From: ted@otherhost Call-Id: 12345@otherhost Cseq: 1 INVITE 44
Call Answer
IP
UAS
UAC
Two-way media flow
Accounting
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Call Release
1a. BYE sip:bob@host SIP/2.0 To: sip:bob@host From: ted@otherhost Call-id: 12345@otherhost Cseq: 1 BYE
Proxy Server
IP
UAS
1b. SIP/2.0 200 OK To:sip:bob@host From: ted@otherhost Call-Id: 12345@otherhost Cseq: 1 BYE
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Vinodworkstation
[email protected]
(1) INVITE vinod @mumbai.tcs.co.in (4) 301 moved permanently Contact:vonod@vinodworkstation. mumbai.tcs.co.in (5) ACK [email protected]
Vinod@vinodworkstation
anilworkstation
Redirect Server
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SIP Interoperability
In the era of network convergence, a key challenge for the network operators and serviced providers is how to ensure interoperability between different communications protocol SIP is fully capable of interoperating with
H.323 ISUP (PSTN/ISDN Network)
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?
PSTN
?
PSTN User
PSTN User
?
SIP User 50
Advantage:
- Both use RTP/RTCP for media transfer SIP TCP/UDP IP
IP IP Network Network
IP IP Network Network
?
H.323 User SIP User NITIN GARG, SDE(5ESS), ALTTC 51
IP IP Network Network
IP IP Network Network
H.323 User
Message Mapping Address mapping Call sequence mapping Audio/video formats mapping Media Switching Function (MSF) SIP User
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SIP-T
SIP-T = Session Initiation Protocol Telephony
previously referred to as SIP+ or SIP BCP-T a collection of internet drafts that extend SIP to support inter-Media Gateway Controller (MGC) communications. SIP-T is an interface agreement on a collection of standards as opposed to a separate protocol
SIP-T describes how to interwork SIP and ISUP SIP-T directly negotiates a media connection between gateways. Endpoint information is carried in SDP (Session Description Protocol) which can describe both IP and ATM endpoints.
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SIP-T purpose
Proxy
IP network
SS7 network
Proxy Proxy
SS7 network
PSTN to IP interworking
MGC
SIP agent
IP network
SS7 network
Proxy
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SIP-PSTN Interworking
Convert from ISUP to SIP SIP Proxy SIP TCP/UDP IP
ISUP MTP
SGW
ISUP IP
IP IP Network Network
SS7 SS7
MGC performs Interworking SIGTRAN
PSTN PSTN
PSTN User SIP User
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SIP-PSTN Interworking
IAM 1 1 2 ACM 2 3 ANM 3 Ringing Response OK Response INVITE Request Trying Response 1
ISUP Message
MGC
IP IP
4 SIP Client
PSTN PSTN
PCM REL 6 NITIN GARG, SDE(5ESS), ALTTC RLC 5 6
SIP - Bridging
PSTN PSTN
PSTN PSTN
ISUP
PSTN PSTN
PSTN PSTN
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ISUP
ISUP
PSTN PSTN
PSTN PSTN
SIP Message Body ISUP Message SIP Header ISUP Message ISUP Message
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