Session Initiation Protocol (SIP) Sess o o o oco (S ) Part II - - PowerPoint PPT Presentation

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Session Initiation Protocol (SIP) Sess o o o oco (S ) Part II Prof. Ai-Chun Pang Graduate Institute of Networking and Multimedia, Dept. of Comp. Sci. and Info. Engr., National Taiwan University Email: acpang@csie.ntu.edu.tw

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SLIDE 1

Session Initiation Protocol (SIP) Sess o

  • oco (S

) Part II

  • Prof. Ai-Chun Pang

Graduate Institute of Networking and Multimedia,

  • Dept. of Comp. Sci. and Info. Engr.,

National Taiwan University Email: acpang@csie.ntu.edu.tw http://www.csie.ntu.edu.tw/~acpang

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SLIDE 2

Examples of SIP Message Sequences Examples of SIP Message Sequences

Via: From: and To: Call-ID:

host-specific host specific

Contact: (for future SIP

message transmission)

Content Length: Content-Length:

Zero, no msg body

CSeq:

A response to any request

must use the same value of CSeq as used in the request.

Expires: Expires:

TTL 0, unreg 2 2011/3/28

  • Prof. Ai-Chun Pang, NTU
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SLIDE 3

Invitation Invitation

A two-party call

Subject:

  • ptional

Content Type: Content-Type:

application/sdp

A dialog ID

T

  • identify a peer-to-peer

relationship between two user agents

Tag in From Tag in From Tag in T

  • Call-ID

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SLIDE 4

Termination of a Call Termination of a Call

CSeq has changed.

Boss< sip:manager@station2.work.com> Daniel< sip:collins@station1.work.com>

BYE sip:manager@station2.work.com SI P/ 2.0 Via: SI P/ 2.0/ UDP station1.work.com Max Forwards: 70

a

Max-Forwards: 70 From: Daniel< sip:collins@work.com> ; tag= 44551 To: Boss< sip:manager@work.com> ; tag= 11222 Call-I D: 123456@station1.work.com CSeq: 2 BYE q Content-Length: 0

b

SI P/ 2.0 200 OK Via: SI P/ 2.0/ UDP station1.work.com Via: SI P/ 2.0/ UDP station1.work.com From: Daniel< sip:collins@work.com> ; tag= 44551 To: Boss< sip:manager@work.com> ; tag= 11222 Call-I D: 123456@station1.work.com CSeq: 2 BYE Content Length: 0

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Content-Length: 0

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SLIDE 5

Redirect Server Redirect Server

An alternative address

302 M d T il

302, Moved T

emporarily

Another INVITE

Same Call-ID Same Call ID CSeq ++

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SLIDE 6

Proxy Server [1/2] Proxy Server [1/2]

Sits between a user-agent client and the far-end user-agent

server

Numerous proxies can reside in a chain between the caller and

ll callee.

The most common scenario will have at least two proxies: one at the

caller and one at the callee end.

It is likely that only the last proxy in the chain changes the Request-URI. The other proxies in the chain would simply use the domain part of the

d R URI d h h received Request-URI to determine the next hop.

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SLIDE 7

Proxy Server [2/2] Proxy Server [2/2]

Via:

The path taken by a request Loop detected, 482 (status code)

The response finds its way back to the originator of the request The response finds its way back to the originator of the request.

Branch: used to distinguish between multiple responses to the same

request

Forking Proxy: Issue a single request to multiple destinations 2011/3/28

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SLIDE 8

Proxy State [1/2] Proxy State [1/2]

Can be either stateless or stateful If stateless, the proxy takes an incoming request,

performs whatever translation and forwards the p corresponding outgoing request and forgets anything.

Retransmission takes the same path (no change on

p ( g retransmission).

If stateful, the proxy remembers incoming requests

, p y g q and corresponding outgoing request.

The proxy is able to act more intelligently on subsequent

p y g y q requests and responses related to the same session.

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SLIDE 9

Proxy State [2/2] Proxy State [2/2]

Record-Route: and Route: Headers

The subsequent requests may not pass through the same path

as the initial request/response.

E g use Contact: E.g., use Contact:

A Proxy might require that it remains in the signaling path for

all subsequent requests to provide some advanced service.

In particular for a stateful proxy

Insert its address into the Record-Route: header The response includes the Record Route: header The response includes the Record-Route: header The information contained in the Record-Route: header is used

in the subsequent requests related to the same call. q q

The Route: header is used to record the path that the request

is enforced to pass.

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SLIDE 10

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SLIDE 11

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  • Prof. Ai-Chun Pang, NTU
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SLIDE 12

Forking Proxy Forking Proxy

A proxy can “fork” requests A user is registered at several locations

;branch=xxx

In order to handle such forking, a proxy must be stateful.

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SLIDE 13

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SLIDE 14

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SLIDE 15

SIP Extensions and Enhancements SIP Extensions and Enhancements

RFC 2543, March 1999 RFC 3261, June 2002

SIP attracts enormous interest. Traditional telecommunications companies, cable TV providers

and ISP

A large number of extensions to SIP have been

proposed. p p

SIP will be enhanced considerably before it becomes an

Internet standard.

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SLIDE 16

183 Session Progress 183 Session Progress

It has been included within the revised SIP spec.

T

  • open one-way audio path from called end to calling end

Enable in-band call progress information to be transmitted

T

  • nes or announcements

Interworking with SS7 network

ACM (Address Complete Message) ACM (Address Complete Message) For SIP-PSTN-SIP connections

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SLIDE 17

INFO Method INFO Method

Be specified in RFC 2976 For transferring information during an ongoing

session

DTMF digits, account-balance information, mid-call signaling

information (from PSTN)

Application-layer information could be transferred in the

middle of a call.

A powerful, flexible tool to support new services

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SLIDE 18

Event Notification Event Notification

Several SIP-based applications have been devised based on the

f b i i f d f concept of a user being informed of some event.

E.g., Instant messaging

RFC 3265 h

dd d th i f t tifi ti

RFC 3265 has addressed the issue of event notification.

SUBSCRIBE and NOTIFY The Event header

Subscriber Notifier

The Event header

SUBSCRIBE a 200 OK NOTIFY b c Current state information 200 OK NOTIFY d e Updated state information

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200 OK f

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SLIDE 19

Instant Messaging Instant Messaging

The IETF working group – SIP for Instant Messaging

and Presence Leveraging Extensions (SIMPLE)

A new SIP method – MESSAGE

This request carries the actual message in a message body.

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SLIDE 20

Daniel<sip:Collins@station1.work.com> Boss<sip:Manager@pc1.home.com> sip:Server.work.com MESSAGE sip:Collins@work com SIP/2 0 a MESSAGE sip:Collins@work.com SIP/2.0 Via: SIP/2.0/UDP pc1.home.net Max-Forwards: 70 From: Boss<sip:Manager@home.net> T

  • : Daniel<sip:Collins@work.com>

MESSAGE sip:Collins@station1.work.com SIP/2.0 Via: SIP/2.0/UDP server.work.com Via: SIP/2.0/UDP pc1.home.net Max-Forwards: 69 b Call-ID: 123456@pc1.home.net CSeq: 1 MESSAGE Content-Type: text/plain Content-Length: 19 Content-Disposition: render From: Boss<sip:Manager@home.net> T

  • : Daniel<sip:Collins@work.com>

Call-ID: 123456@pc1.home.net CSeq: 1 MESSAGE Content-Type: text/plain p

  • Hello. How are you?

Co te t ype: te t/p a Content-Length: 19 Content-Disposition: render

  • Hello. How are you?

SIP/2.0 200 OK Via: SIP/2.0/UDP server.work.com Via: SIP/2.0/UDP pc1.home.net From: Boss<sip:Manager@home.net> SIP/2.0 200 OK Via: SIP/2.0/UDP pc1.home.net From: Boss<sip:Manager@home net> c d T

  • : Daniel<sip:Collins@work.com>

Call-ID: 123456@pc1.home.net CSeq: 1 MESSAGE Content-Length: 0 From: Boss<sip:Manager@home.net> T

  • : Daniel<sip:Collins@work.com>

Call-ID: 123456@pc1.home.net CSeq: 1 MESSAGE Content-Length: 0

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SLIDE 21

Daniel<sip:Collins@station1.work.com> Boss<sip:Manager@pc1.home.com> sip:Server.work.com MESSAGE sip:Manager@home net SIP/2 0 e MESSAGE sip:Manager@home.net SIP/2.0 Via: SIP/2.0/UDP station1.work.com Max-Forwards: 70 From: Daniel<sip:Collins@work.com> T

  • : Boss<sip:Manager@home.net>

MESSAGE sip:Manager@pc1.home.net SIP/2.0 Via: SIP/2.0/UDP server.work.com; Via: SIP/2.0/UDP station1.work.com Max-Forwards: 69 f Call-ID: 456789@station1.work.com CSeq: 1101 MESSAGE Content-Type: text/plain Content-Length: 22 Content-Disposition: render From: Daniel<sip:Collins@work.com> T

  • : Boss<sip:Manager@home.net>

Call-ID: 456789@station1.work.com CSeq: 1101 MESSAGE Content Type: text/plain p I’m fine. How are you? Content-Type: text/plain Content-Length: 22 Content-Disposition: render I’m fine. How are you? SIP/2.0 200 OK Via: SIP/2.0/UDP server.work.com Via: SIP/2.0/UDP station1.work.com From: Daniel<sip:Collins@work com> SIP/2.0 200 OK Via: SIP/2.0/UDP station1.work.com From: Daniel<sip:Collins@work com> g h From: Daniel<sip:Collins@work.com> T

  • : Boss<sip:Manager@home.net>

Call-ID: 456789@station1.work.com CSeq: 1101 MESSAGE Content-Length: 0 From: Daniel<sip:Collins@work.com> T

  • : Boss<sip:Manager@home.net>

Call-ID: 456789@station1.work.com CSeq: 1101 MESSAGE Content-Length: 0

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SLIDE 22

REFER Method REFER Method

To enable the sender of the request to instruct the receiver to

hi d contact a third party

With the contact details for the third party included within the REFER request For Call Transfer applications For Call Transfer applications

The Refer-to: and Refer-by: Headers The dialog between Mary and Joe remains established The dialog between Mary and Joe remains established.

Joe could return to the dialog after consultation with Susan. 22 2011/3/28

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SLIDE 23

sip:Mary@station1.work.com sip:Joe@station2.work.com sip:Susan@station3.work.com a REFER sip:Joe@station2.work.com SIP/2.0 Via: SIP/2.0/UDP station1.work.com Max-Forwards: 70 From: Mary<sip:Mary@work.com>; tag=123456 T J < i J @ k > t =67890 a T

  • : Joe<sip:Joe@work.com>; tag=67890

Contact: Mary<Mary@station1.work.com> Refer-T

  • : Susan<sip:Sussan@station3.work.com>

Call-ID: 123456@station1.work.com CSeq: 123 REFER Content-Length: 0 SIP/2.0 202 Accepted Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK789 From: Mary<sip:Mary@work.com>; tag=123456 T J i J @ k 67890 INVITE sip:Susan@station3.work.com SIP/2.0 Via: SIP/2.0/UDP station2.work.com; b c T

  • : Joe<sip:Joe@work.com>; tag=67890

Contact: Joe<Joe@station2.work.com> Call-ID: 123456@station1.work.com CSeq: 123 REFER Content-Length: 0 Max-Forwards: 70 From: Joe<sip:Joe@work.com>; tag=abcxyz T

  • : Susan<sip:Susan@station3.work.com>

Contact: Joe<Joe@station2.work.com> Call-ID: 67890@station2.work.com g Call ID: 67890@station2.work.com CSeq: 567 INVITE Content-Type: application/sdp Content-Length: xx Content-Disposition: session {message body}

23

{message body}

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SLIDE 24

sip:Mary@station1.work.com sip:Joe@station2.work.com sip:Susan@station3.work.com e SIP/2.0 200 OK Via: SIP/2.0/UDP station2.work.com From: Joe<sip:Joe@work.com>; tag=abcxyz T

  • : Susan<sip:Susan@station3 work com>; tag=123xyz

T

  • : Susan<sip:Susan@station3.work.com>; tag 123xyz

Call-ID: 67890@station2.work.com CSeq: 567 INVITE Content-Type: application/sdp Content-Length: xx C t t Di iti i f g Content-Disposition: session {message body} ACK sip:Susan@station3.work.com SIP/2.0 Via: SIP/2.0/UDP station2.work.com NOTIFY sip:Mary@station1.work.com SIP/2.0 Via: SIP/2.0/UDP station2.work.com Max-Forwards: 70 Via: SIP/2.0/UDP station2.work.com Max-Forwards: 70 From: Joe<sip:Joe@work.com>; tag=abcxyz T

  • : Susan<sip:Susan@station3.work.com>; tag=123xyz

Call-ID: 67890@station2.work.com CS 567 ACK Max-Forwards: 70 From: Joe<sip:Joe@work.com> T

  • : Mary<sip:Mary@work.com>

Contact: Joe<Joe@station2.work.com> Call-ID: 123456@station1.work.com CS 124 NOTIFY CSeq: 567 ACK Content-Length: 0 CSeq: 124 NOTIFY Content-Type: message/sipfrag;version=2.0 Content-Length: 15 SIP/2.0 200 OK h SIP/2.0 200 OK Via: SIP/2.0/UDP station2.work.com From: Joe<sip:Joe@work.com> T

  • : Mary<sip:Mary@work.com>

C ll ID 123456@ t ti 1 k h

24

Call-ID: 123456@station1.work.com CSeq: 124 NOTIFY Content-Length: 0

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SLIDE 25

Reliability of Provisional Responses [1/2] Reliability of Provisional Responses [1/2]

Provisional Responses

100 (trying), 180 (ringing), 183 (session in progress) Are not answered with an ACK

If th

i t UDP

If the messages is sent over UDP

Unreliable

Lost provisional response may cause problems when Lost provisional response may cause problems when

interoperating with other network

183 (to create a one-way audio path)

( y p )

The state machine of a PSTN switch is driven by some of provisional

responses.

E.g., a call to an unassigned number

If the provisional response “183” is lost, the caller might left in the dark and

not understand why the call did not connect.

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SLIDE 26

Reliability of Provisional Responses [2/2] Reliability of Provisional Responses [2/2]

RFC 3262

Reliability of Provisional

ClientA@network.com ServerB@network.com

Reliability of Provisional

Responses in SIP

Supported: 100rel RSeq Header

a

RSeq Header

Response Seq +1, when retxm

INVITE sip:ServerB@network.com SIP/2.0 Via: SIP/2.0/UDP ClientA.network.com Supported: 100rel Require: 100rel From: sip:ClientA@network.com; tag=lmnop123 To: sip:ServerB@network com

PRACK

  • Prov. Resp. ACK

RAck Header

To: sip:ServerB@network.com Call-ID: 123456@ClientA.network.com CSeq: 1 INVITE

??

SIP/2.0 180 Ringing b

RAck Header

Response ACK In PRACK RSeq+CSeq

g g Via: SIP/2.0/UDP ClientA.network.com Require: 100rel RSeq: 567890 From: sip:ClientA@network.com; tag=lmnop123 To: sip:ServerB@network.com; tag = xyz123 Call-ID: 123456@ClientA network com Response Lost Response Retransmit

RSeq+CSeq

Should not

Apply to 100

Call-ID: 123456@ClientA.network.com CSeq: 1 INVITE c SIP/2.0 180 Ringing Via: SIP/2.0/UDP ClientA.network.com Require: 100rel

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Default timer value = 0.5 s

...

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SLIDE 27

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SLIDE 28

SDP Inclusion in SIP Messages

Daniel<sip:Collins@station1.work.com> Boss<sip:Manager@station2.work.com> INVITE sip:Manager@station2.work.com SIP/2.0 From: Daniel<sip:Collins@station1.work.com>; tag = abcd1234 a T

  • : Boss<sip:Manager@station2.work.com>

CSeq: 1 INVITE Content-Length: 213 Content-Type: application/sdp Content-Disposition: session Content Disposition: session v=0

  • =collins 123456 001 IN IP4 station1.work.com

s= c=IN IP4 station1 work com c=IN IP4 station1.work.com t=0 0 m=audio 4444 RTP/AVP 2 a=rtpmap 2 G726-32/8000 m=audio 4666 RTP/AVP 4 a=rtpmap 4 G723/8000 m=audio 4888 RTP/AVP 15 a=rtpmap 15 G728/8000 SIP/2.0 200 OK

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b SIP/2.0 200 OK …

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SLIDE 29

Daniel<sip:Collins@station1.work.com> Boss<sip:Manager@station2.work.com> b SIP/2.0 200 OK From: Daniel<sip:Collins@station1.work.com>; tag = abcd1234 T

  • : Boss<sip:Manager@station2.work.com>; tag = xyz789

CSeq: 1 INVITE Content-Length: 163 g Content-Type: application/sdp Content-Disposition: session v=0

  • =collins 45678 001 IN IP4 station2 work com
  • =collins 45678 001 IN IP4 station2.work.com

s= c=IN IP4 station2.work.com t=0 0 m=audio 0 RTP/AVP 2 di 0 RTP/AVP 4 m=audio 0 RTP/AVP 4 m=audio 6666 RTP/AVP 15 a=rtpmap 15 G728/8000 c ACK sip:manager@station2.work.com SIP/2.0 i i C i @ i 234 d From: Daniel<sip:Collins@station1.work.com>; tag = abcd1234 T

  • : Boss<sip:Manager@station2.work.com>; tag = xyz789

CSeq: 1 ACK Content-Length: 0 d Conversation

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SLIDE 30

Integration of SIP Signaling and Resource Management Management

Ensuring that sufficient resources are available to handle a media

t i i t t stream is very important.

T

  • provide a high-quality service for a carrier-grade network

The signaling might take a different path from the media The signaling might take a different path from the media.

The successful transfer of signaling messages does not imply to a

successful transfer of media.

Assume resource-reservation mechanisms are available

On a per-session basis

End-to-end network resources are reserved as part of session establishment.

On an aggregate basis

A certain amount of network resources are reserved in advance for a certain A certain amount of network resources are reserved in advance for a certain

type of usage.

Policing functions at the edge of the network 2011/3/28

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SLIDE 31

Session Establishment with End-to-End Resource Reservation Resource Reservation

Reserving network resources in

advance of altering the called user

A IETF document →“Integration of

Resource Management and SIP” Resource Management and SIP

By using the provisional responses

and UPDATE method

By involving extensions to SDP

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SLIDE 32

Example of Aggregate based Reservation Example of Aggregate-based Reservation

Each participant deals with

network access permission at its network access permission at its

  • wn end.

Mandatory means that the session

can not continue unless the required resources are definitely available.

None is the initial situation and

indicates that no effort to reserve h t t k l resources has yet taken place.

Response 580 (precondition

failure)

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SLIDE 33

Usage of SIP for Features/Services [1/2] Usage of SIP for Features/Services [1/2]

Call-transfer application (with REFER method) Personal mobility through the use of registration One number service through forking proxy Call-completion services by using Retry-After: header SIP address is a URL

Click-to-call applications

The existing supplementary services in traditional telephony

Call transfer, call waiting, call forwarding, multi-party calling, call screening

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SLIDE 34

Usage of SIP for Features/Services [2/2] Usage of SIP for Features/Services [2/2]

Proxy invokes various types of advanced feature logic.

Policy server (call-routing, QoS) Authentication server Application server Application server

The network can use the Parley Open Service Access (OSA)

approach, utilizing application programming interfaces (APIs) pp , g pp p g g ( ) between the nodes.

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SLIDE 35

Call Forwarding Call Forwarding

On busy

User1 sip:Server.work.com User2 User3

486, busy here With the same To, User 3 can

recognize that this call is a

I NVI TE sip:user2@server.work.com SI P/ 2.0 From: sip:user1 To: sip:user2@work.com

recognize that this call is a forwarded call, originally sent to User 2.

C

t t h d i 200

To: sip:user2@work.com Contact: User1 CSeq: 1 I NVI TE SI P/ 2.0 100 Trying From: sip:user1 To: sip:user2@work.com CSeq: 1 I NVI TE I NVI TE sip:user2@server.work.com SI P/ 2.0 From: sip:user1 To: sip:user2@work com

Contact: header in 200 response Call-forwarding-on-no-answer Timeout

CSeq: 1 I NVI TE To: sip:user2@work.com Contact: User1 CSeq: 1 I NVI TE SI P/ 2.0 486 Busy Here From: sip:user1 To: sip:user2@work.com CSeq: 1 I NVI TE

Timeout CANCEL method

CSeq: 1 I NVI TE I NVI TE sip:user3@server.work.com SI P/ 2.0 From: sip:user1 To: sip:user2@work.com CSeq: 2 I NVI TE SI P/ 2.0 200 OK From: sip:user1 To: sip:user2@work.com Contact: sip:user3@work.com CSeq: 2 I NVI TE SI P/ 2.0 200 OK From: sip:user1 To: sip:user2@work.com Contact: sip:user3@work.com CSeq: 1 I NVI TE

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CSeq: 1 I NVI TE

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SLIDE 36

Consultation Hold Consultation Hold

A SIP UPDATE User A asks User B a question,

and User B needs to check with U C f th t User C for the correct answer.

If User C needs to talk to User A

directly User B could use the directly, User B could use the REFER method to transfer the call to User C.

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SLIDE 37

PSTN Interworking PSTN Interworking

  • PSTN Interworking
  • A SIP URL to a telephone

Manager@work.com Proxy.work.com PSTN switch NGW

  • A SIP URL to a telephone

number

  • A network gateway
  • Seamless interworking

I NVI TE a b 100 (Trying)

  • Seamless interworking

between two different protocols is not quite easy.

  • One-to-one mapping

100 (Trying) I NVI TE 100 (Trying) I AM c d e f

  • One to one mapping

between these protocols

  • PSTN – SIP – PSTN
  • MIME media types

183 (Session Progress) Session description One-way audio g i 183 (Session Progress) Session description ACM One-way audio h

  • MIME media types
  • SIP for T

elephony (SIP-T)

  • The whole issue of

interworking with SS7 is

One-way audio j One-way audio ANM 200 (OK) Updated session description 200 (OK) k l

interworking with SS7 is fundamental to the success of VoIP in the real world.

m Updated session description ACK ACK n

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Two-way audio Two-way audio

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SLIDE 38

Summary Summary

Signaling in VoIP networks

Simple, yet flexible Easier to implement Fit and coexist well with the PSTN

SIP is the protocol of choice for the evolution of next-

p generation wireless networks.

SIP-based mobile devices become available. SIP-based network elements are introduced within mobile

networks.

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