Session Initiation Protocol (SIP)

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Session Initiation Protocol (SIP)

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SIP Extensions and Enhancements

n RFC 2543, March 1999

n

RFC 3261, June 2002

n

SIP has attracted enormous interest.

n

Traditional telecommunications companies, cable TV providers and ISP

n A large number of extensions to SIP have been proposed.

n

SIP will be enhanced considerably before it

becomes an Internet standard.

(3)

183 Session Progress

n It has been included within the revised SIP spec.

n

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

n

Enable in-band call progress information to be transmitted

n

Tones or announcements

n

Interworking with SS7 network

n

ACM (Address Complete Message)

n

For SIP-PSTN-SIP connections

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The Supported Header

n The Base RFC 2543

n

The Require: Header

n

In request (client ->server)

n

A client indicates that a server must support certain extension.

n

The Unsupported Header

n

In response (server -> client)

n

420 (bad extension)

n

A cumbersome way of determining what extensions a server does or does not support

n The Supported: Header (RFC 3261)

n

May be included in OPTIONS request

n

Associated with the Supported: header is 421 (extension required) response.

n

Can also be included in responses

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SIP INFO Method

n Be specified in RFC 2976

n For transferring information during an ongoing session

n

DTMF digits, account-balance information, mid-call signaling information (from PSTN)

n

Application-layer information could be transferred in the middle of a call.

n A powerful, flexible tool to support new

services

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SIP Event Notification

n Several SIP-based

applications have been devised based on the concept of a user being informed of some event.

n

E.g., Instant messaging

n RFC 3265 has addressed the issue of event

notification.

n

SUBSCRIBE and NOTIFY

n

The Event header

Subscriber Notifier

SUBSCRIBE

200 OK NOTIFY 200 OK

NOTIFY 200 OK

a b c d e f

Current state information

Updated state information

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SIP for Instant Messaging

n The IETF working group – SIP for Instant

Messaging and Presence Leveraging Extensions (SIMPLE)

n A new SIP method – MESSAGE

n

This request carries the actual message in a message body.

n

A MESSAGE request does not establish a SIP dialog.

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Daniel<sip:Collins@station1.work.com>

Boss<sip:Manager@pc1.home.com> sip:Server.work.com

MESSAGEsip:Collins@work.com SIP/2.0 Via: SIP/2.0/UDP pc1.home.net;

branch=z9hG4bK7890 Max-Forwards: 70

From: Boss<sip:Manager@home.net>

To: Daniel<sip:Collins@work.com>

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

Content-Type: text/plain Content-Length: 19

Content-Disposition: render Hello. How are you?

MESSAGE sip:Collins@work.com SIP/2.0 Via: SIP/2.0/UDP server.work.com;

branch=z9hG4bKxyz1

Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890 Max-Forwards: 69

From: Boss<sip:Manager@home.net>

Content-Disposition: render Hello. How are you?

SIP/2.0 200 OK

Via: SIP/2.0/UDP server.work.com;

branch=z9hG4bKxyz1

Via: SIP/2.0/UDP pc1.home.net; branch=z9hG4bK7890 From: Boss<sip:Manager@home.net>

Content-Length: 0 SIP/2.0 200 OK

Via:SIP/2.0/UDP pc1.home.net;branch=z9hG4bK7890 From: Boss<sip:Manager@home.net>

Content-Length: 0 ab

c d

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Daniel<sip:Collins@station1.work.com>

Boss<sip:Manager@pc1.home.com> sip:Server.work.com

MESSAGE sip:Manager@home.net SIP/2.0 Via: SIP/2.0/UDP station1.work.com;

branch=z9hG4bK123 Max-Forwards: 70

From: Daniel<sip:Collins@work.com>

To: Boss<sip:Manager@home.net>

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

Content-Disposition: render I’m fine. How are you?

MESSAGEsip:Manager@home.net SIP/2.0

Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKabcd

Via: SIP/2.0/UDP station1.work.com; branch=z9hG4bK123

Max-Forwards: 69

Content-Disposition: render I’m fine. How are you?

SIP/2.0 200 OK

Via: SIP/2.0/UDP server.work.com; branch=z9hG4bKabcd

Content-Length: 0

SIP/2.0 200 OK

Via: SIP/2.0/UDP station1.work.com;

branch=z9hG4bK123

Content-Length: 0 ef

g h

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SIP REFER Method

n To enable the sender of the request to instruct the receiver to contact a third party

n

With the contact details for the third party included within the REFER request

n

For Call Transfer applications

n The Refer-to: and Refer-by: Headers

n The dialog between Mary and Joe remains established.

n

Joe could return to the dialog after consultation with Susan.

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sip:Mary@station1.work.co

m sip:Joe@station2.work.com sip:Susan@station3.work.com

REFER sip:Joe@station2.work.com SIP/2.0

From: Mary<sip:Mary@work.com>; tag=123456 To: Joe<sip:Joe@work.com>; tag=67890

Contact: Mary<Mary@station1.work.com>

Refer-To: Sussan<sip:Sussan@station3.work.com>

Call-ID: 123456@station1.work.com CSeq: 123 REFER

Content-Length: 0 SIP/2.0 202 Accepted

From: Mary<sip:Mary@work.com>; tag=123456 To: 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

INVITE sip:Susan@station3.work.com SIP/2.0

Via: SIP/2.0/UDP station2.work.com; branch=z9hG4bKxyz1

From: Joe<sip:Joe@work.com>; tag=abcxyz To: Susan<sip:Susan@station3.work.com>

Contact: Joe<Joe@station2.work.com>

Call-ID: 67890@station2.work.com CSeq: 567 INVITE

Content-Type: application/sdp Content-Length: xx

Content-Disposition: session {message body}

a

b c

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sip:Mary@station1.work.co m

sip:Joe@station2.work.com sip:Susan@station3.work.com

e

f g

SIP/2.0 200 OK

From: Joe<sip:Joe@work.com>; tag=abcxyz

To: Susan<sip:Susan@station3.work.com>; tag=123xyz Call-ID: 67890@station2.work.com

CSeq: 567 INVITE

Content-Type: application/sdp Content-Length: xx

Content-Disposition: session {message body}

ACKsip:Susan@station3.work.com SIP/2.0

From: Joe<sip:Joe@work.com>; tag=abcxyz

To: Susan<sip:Susan@station3.work.com>; tag=123xyz Call-ID: 67890@station2.work.com

CSeq: 567 ACK Content-Length: 0 NOTIFY sip:Mary@station1.work.com SIP/2.0

Via:SIP/2.0/UDP station2.work.com;branch=z9hG4bK123

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

From: Mary<sip:Mary@work.com>; tag=123456 Contact: Joe<Joe@station2.work.com>

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

Content-Type: message/sipfrag;version=2.0 Content-Length: 15

SIP/2.0 200 OK

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

From: Mary<sip:Mary@work.com>; tag=123456 Call-ID: 123456@station1.work.com

CSeq: 124 NOTIFY Content-Length: 0 h

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Reliability of Provisional Responses [1/2]

n Provisional Responses

n

100 (trying), 180 (ringing), 183 (session in progress)

n

Are not answered with an ACK

n If the messages is sent over UDP

n

Unreliable

n Lost provisional response may cause problems when interoperating with other network

n

180, 183 → Q.931 alerting or ISUP ACM

n

To drive a state machine

n

E.g., a call to an unassigned number

n

ACM to create a one-way path to relay an announcement such as

“The number you have called has been changed”

n

If the provisional response is lost, the called might left in the dark

and not understand why the call did not connect.

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n

RFC 3262

n

Reliability of Provisional Responses in SIP

n

Supported: 100rel

n

RSeq Header

n

Response Seq

n

+1, when retxm

n

RAck Header

n

Response ACK

n

In PRACK

n

RSeq+CSeq

n

PRACK

n

Prov. Resp. ACK

n

Should not

n

Apply to 100

n

Default timer value = 0.5 s

Reliability of Provisional Responses [2/2]

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The SIP UPDATE Method

n To enable the modification of session

information before a final response to an INVITE is received

n

The dialog is in the early state (An INVITE that receives a 183 response that includes a message body)

n

The message body might establish a media stream from callee to caller for sending a ring tone or music while the called party is alerted.

n

The UPDATE method can be used to change the codec

n Another important usage is when reserving

network resources as part of a SIP session

establishment.

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Integration of SIP Signaling and Resource Management [1/2]

n

Ensuring that sufficient resources are available to handle a media stream is very important.

n

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

n

The signaling might take a different path from the media.

n

The successful transfer of signaling messages does not imply to a successful transfer of media.

n

Assume resource-reservation mechanisms are available (Chapter 8)

n

On a per-session basis

n

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

n

On an aggregate basis

n

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

n

Policing functions at the edge of the network

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Integration of SIP Signaling and Resource Management [2/2]

n Reserving network

resources in advance of altering the called user

n A new draft –

“Integration of Resource Management and SIP”

n

By using the provisional responses and UPDATE method

n

By involving extensions to

SDP

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Example of e2e Resource Reservation [1/2]

n SDP for initial INVITE

v=0 o=userA 45678 001 IN IP4 stationA.network.com s= c=IN IP4 stationA.nework.com

t=0 0

m=audio 4444 RTP/AVP 0 a=curr: qos e2e none

a=des: qos mandatory e2e sendrecv

n SDP for 183 response

v=0 o=userB 12345 001 IN IP4 stationB.network.com s= c=IN IP4 stationB.nework.com

t=0 0

m=audio 6666 RTP/AVP 0 a=curr: qos e2e none

a=des: qos mandatory e2e sendrecv

a=conf: qos e2e recv

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Example of e2e Resource Reservation [2/2]

n SDP for UPDATE

v=0 o=userA 45678 001 IN IP4 stationA.network.com s= c=IN IP4 stationA.nework.com

t=0 0

m=audio 4444 RTP/AVP 0 a=curr: qos e2e send

a=des: qos mandatory e2e sendrecv

n SDP for 200 response

v=0 o=userB 12345 001 IN IP4 stationB.network.com s= c=IN IP4 stationB.nework.com

t=0 0

m=audio 6666 RTP/AVP 0 a=curr: qos e2e sendrecv

a=des: qos mandatory e2e sendrecv

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Example of Aggregate- based Reservation

n

Each participant deals with network access permission at its own end.

n

Mandatory means that the session can not continue unless the required resources are

definitely available.

n

None is the initial situation and indicates that no effort to

reserve resources has yet taken place.

n

Response 580 (precondition

failure)

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

n Call-transfer application (with REFER method)

n Personal Mobility through the use of registration

n One number service through forking proxy

n Call-completion services by using Retry-After: header

n To carry MIME content as well as an SDP description

n

To include a piece of text, an HTML document, an image and so on

n SIP address is a URL

n

Click-to-call applications

n The existing supplementary services in traditional telephony

n

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

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

n

Proxy invokes various types of advanced feature logic.

n

Policy server (call-routing, QoS)

n

Authentication server

n

Use the services of an IN SCP over INAP

n

The network might use the Parley Open Service Access (OSA) approach, utilizing application programming

interfaces (APIs) between the nodes.

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Call Forwarding

n

On busy

n

486, busy here

n

With the same To, User 3 can recognize that this call is a forwarded call,

originally sent to User 2.

n

Contact: header in 200 response

n

Call-forwarding-on-no- answer

n

Timeout

n

CANCEL method

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Consultation Hold

n A SIP UPDATE

n User A asks User B a

question, and User B need to check with User C for the correct answer.

n If User C needs to talke to User A directly, User B

could use the REFER

method to transfer the

call to User C.

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PSTN Interworking

n

PSTN Interworking

n

A SIP URL to a telephone number

n

A network gateway

n

Seamless interworking between two different

protocols is not quite easy.

n

One-to-one mapping between these protocols

n

PSTN – SIP – PSTN

n

MIME media types

n

For ISUP

n

SIP for Telephony (SIP-T)

n

The whole issue of

interworking with SS7 is

fundamental to the success

of VoIP in the real world.

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Interworking with H.323

n SIP-H.323 interworking gateway

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Summary

n The future for signaling in VoIP networks

n

Simple, yet flexible

n

Easier to implement

n

Fit well with the media gateway control protocols

n