sip/session mobility 陳旻秀 中華大學資訊工程學系. outline what’s sip sip overview...

SIP/Session mobility

陳旻秀中華大學資訊工程學系

Outline

What’s SIP SIP Overview Session Mobility Split Session

What’s SIP

Session Initiation Protocol

Application-layer control protocol

Main function -User Location Service

-Session establishment

-Session participant management

-Limited feature establishment

What is SIP ?

Session Initiation Protocol MMUSIC working group: RFC 2543 SIP working group: RFC 3261

SIP enable Internet endpoints (User agents) to discover one another and to agree on a characterization of a session they would like to share All kind of real-time multimedia types, such as

voice, video or text messages

Characteristic of SIP

Open - 如同 HTTP 以代碼表示服務狀況並採用和 SMTP

同樣的帳號模式

Extensible

- 可以像 DNS 一樣利用上下游關係進行擴充

RFC

RFC文件編號 說明

RFC2543 第一份 SIP RFC

RFC3261 SIP 的基礎

RFC3262 管理應答關係

RFC3263 SIP Proxy 規則

RFC3264 offer/answer model

RFC3265 事件表示

SIP Function and Property

A signaling protocol (application-layer control protocol) For creating, modifying, and terminating sessions

Separate signaling and media streams

End-to-End transmission

Establishing a sip session within the same domain

SIP Registrar Server

SIP Proxy Server

SIP User Agent ServerSIP User Agent Client

32

線條說明

Request

Response

Domain A

1

6

4

5

RTP

SIP Architecture

request

response

non-SIP protocol

SIP User Agent Client(Caller)

SIP User Agent Server(Callee)

SIP Proxy SIP Proxy

SIP Proxy

Location Server

SIP Redirect Server

1

2

3

4

5

6

78

9

121314

RTP

1011

Establishing a sip sessionin dissimilar domain

Domain A SIP Proxy Server

Domain A SIP Registrar Server

Domain B SIP Proxy Server

Domain B SIP Registrar Server

User Agent ClientUser Agent Server

SIP Redirect Server

2 3

4

9

1018 7

65

線條說明

Request

Response

RTP

Domain A Domain B

Network Entities

UA (User Agent) UAC (User agent client)

Initiate SIP request UAS (User agent server)

Receive request, return a response

SIP Proxy Server Forwards requests to the next any kind of SIP

servers or UA

Network Entities (cont.)

SIP Redirect Server Map SIP address into new address and return to

the client SIP Registrar Server

Accept user register requests to update user address location database

Location Server A database to obtain information about a callee's

possible location

Registrar Server

Location ServerRegistrar

Proxy

UA

Register

Store

Query Response

SIP Addressing

SIP URI E-mail like user@host sip:Alice@taipei.com

Request Method

INVITE The user is begin invited to participate in a session.

ACK The client has received a final response to an INVITE.

OPTIONS The server is begin queried as to its capabilities.

BYE The user wishes to release the call.

CANCEL It cancels a pending request (not completed request).

REGISTER

It conveys the user’s location information to a SIP server.

Response Code

1xx --- Provisional 100 Trying

2xx --- Successful 200 OK

3xx --- Redirection 302 Moved Temporarily

4xx --- Request Failure 486 Busy Here

5xx --- Server Failure 501 Not Implemented

6xx --- Global Failure 604 Does Not Exist Anywhere

Basic Call Flow

RTP

UAC (Caller)Alice’s softphone

UAS (Callee)Bob’s SIP phone

taipei.comproxy

tainan.comproxy

INVITE 1INVITE 2

100: Trying

3INVITE 4

100: Trying

5

180: Ringing

6180: Ringing

7180: Ringing

8 200: OK 9

200: OK10

200: OK11

ACK 12

BYE 13

200: OK 14

Busy Call Flow

UAC (Caller)Alice’s softphone

UAS (Callee)Bob’s SIP phone

taipei.comproxy

tainan.comproxy

INVITE 1INVITE 2

100: Trying

3INVITE 4

100: Trying

5

486: Busy

6

486: Busy

8

486: Busy

10

ACK 7

ACK 9

ACK 11

Cancel Call Flow

UAC (Caller)Alice’s softphone

UAS (Callee)Bob’s SIP phone

taipei.comproxy

tainan.comproxy

INVITE 1INVITE 2

100: Trying

3 INVITE 4100: Trying

5180: Ringing

6180: Ringing

7180: Ringing

8

200: OK10

CANCEL9

200: OK10

CANCEL9

200: OK10

CANCEL9

487: RT 11

ACK 12487: RT 11

ACK 12487: RT 11

ACK 12

SIP Message Syntax Overview Text-based

Use the UTF-8 charset (RFC 2279) Syntax similar to HTTP/1.1 (RFC 2616)

generic-message = start-line *message-header CRLF [message-body]

Start Line

Request-Line = Method SP Request-URI SP SIP-Version CRLF

Status-Line = SIP-Version SP Status-Code SP Reason-Phrase CRLF

Ex: INVITE sip:Alice@taipei.com SIP/2.0

Ex: SIP/2.0 200 OK

start-line = Request-line | Status-line

Message Header

General Header Be applied to both request and response messages

Entity Header Define information about the message body

Request Header Allow the client to pass additional information about the

request Response Header

Allow the server to pass additional information about the response

message-header = ( general-header | request-header | response-header | entity-header)

Message Header (cont.)

Basic Call Flow

RTP

UAC (Caller)Alice’s softphone

UAS (Callee)Bob’s SIP phone

taipei.comproxy

tainan.comproxy

INVITE 1INVITE 2

100: Trying

3INVITE 4

100: Trying

5

180: Ringing

6180: Ringing

7180: Ringing

8 200: OK 9

200: OK10

200: OK11

ACK 12

BYE 13

200: OK 14

Message-body

Blank-line

Message-header

An Example of SIP Request

Start-lineINVITE sip:bob@tainan.com SIP/2.0 Via: SIP/2.0/UDP pc33.taipei.com:5060From: Alice <sip:alice@taipei.com> To: Bob <sip:bob@tainan.com> Call-ID: a84b4c76e66710@taipei.com CSeq: 1 INVITE Contact: <sip:alice@100.101.102.103> Content-Type: application/sdp Content-Length: 147 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com s=Session SDP c=IN IP4 100.101.102.103 t=0 0m=audio 49172 RTP/AVP 0 a=rtpmap:0 PCMU/8000

Start-line

Message-header

An Example of SIP Response

SIP/2.0 100 Trying Via: SIP/2.0/UDP pc33.taipei.com:5060From: Alice <sip:alice@taipei.com> To: Bob <sip:bob@tainan.com> Call-ID: a84b4c76e66710@taipei.com CSeq: 1 INVITE Content-Length: 0

If no body in message, there is no Content-Type and Content-Length = 0

SIP Extensions

SIP has attracted enormous interest Simple Easy to implement Easy to add functions

A large number of extensions to SIP has been proposed INFO Method REFER Method

SIP INFO Method

Specified in RFC 2976 For transferring information during an

ongoing session The transfer of mid-call signaling information

SIP REFER Method

Specified in RFC 3515 To enable the sender of the request to

instruct the receiver to contact a third party For Call Transfer applications

New header fields Refer-to Refer-by

SIP for Instant Messaging

The IETF working group – SIP for Instant Messaging and Presence

Leveraging Extensions (SIMPLE)

Four mobile issue of SIP

Terminal Mobility

Personal Mobility

Session Mobility

Service Mobility

Limitation of mobile IP

Triangle routing

Triangle registration

Encapsulation overhead

Need for home address

ISP must offer home agent service

MIP v.s. SIP

Mobility IP bind a IP address to a temporary care-of address

(ex 140.126.5.2)

SIP bind a user-level identifier to a temporary IP address

(ex sip:ant@ant.csie.chu.edu.tw)

Session Mobility

Session mobility allows a user to maintain a media session even while changing terminals

Alice@mobileAlice@fixed Bob@fixed

pc pc

phone

Personal Mobility

sagual@sip.csie.chu.edu.tw

guest@sip.csie.chu.edu.tw

visiter@sip.csie.chu.edu.tw

xavier@ant.csie.chu.edu.tw

sip.csie.chu.edu.tw

Ant.csie.chu.edu.tw

Service Mobility

Services setting bind with SIP identified “home” server Call Processing Language (CPL)

Contact: Carol <sip:carol@example.com>;language="en,es,de";media="audio,video,application/chat";duplex="full";priority="urgent"

Types of session mobility

Third-party call control

RFC 3725

REFER

RFC 3515

Related work

K. Kaneko, H. Morikawa, and T. Aoyama:"Session Layer Mobility Support for 3C Everywhere Environments," In Proceeding of the 6th International Symposium on Wireless Personal Multimedia Communications (WPMC 2003), vol.2, pp.347-351, Yokosuka, Japan, October 2003.

Migration mechanism

SL Middleware

BSD socket

Port 1234

A

SL Middleware

BSD socket

Port 5678

B

SL Middleware

BSD socket

Port 1357

C

Media Stream

1. Migration Message

2. Create New Socket

3. Notify B

BSD socket

Port 24684. Create New Socket

Media Stream

Related work (cond.)

Ohta, K.; Yoshikawa, T.; Nakagawa, T.; Isoda, Y.; Kurakake, S.; NTT DoCoMo,Inc.“Adaptive terminal middleware for session mobility ” ,Distributed Computing Systems Workshops, 2003. Proceedings. 23rd International Conference on , 2003 Pages:394 - 399

Adaptive terminal middleware Session management Reliable virtual socket Proactive soft handoff

What middle layer can’t do？

Resolution300x400 Resolution

1024x768

Resolution1024x768

A B

C

Resolution 300x400

Media Stream

Resolution 1024x768

Middle layer can’t modify the application layer.

So, if we use middle layer for session mobility, we

must use resolution 300x400 here.

Session Mobility using SIP

H. Schulzrinne and E. Wedland, “Application-layer mobility using SIP”,ACM SIGMOBILE Mobile Computing and Communications Review,Vol. 4, Number 3, July 2000, pp.47-57

RFC 3515 The Session Initiation Protocol (SIP) Refer Method. R. Sparks. April 2003.

draft-ietf-sipping-3pcc-06

REFER Mechanism

B1

B2

A REFER AReferred-By:B1

INVITE AReferred-By:B1

BYE A

1

2

3

Third Party Call Control

Third party call control refers to the ability of one entity to create a call in which communication is actually between other parties

Controller

Party A Party B

Third Party Call Control (Cond.) Third Party Call Control (3pcc) is often used

for operator services and conferencing Click-to-dial allows a user to click on a web

page when they wish to speak to a customer service

Offer/Answer Model

RFC 3264 Two entities can use SDP to arrive at a common view of

a multimedia session between them

UAC (Caller)Offerer

UAS (Callee)Answerer

INVITE

Offer : audio video

200: OK

Answer : audio

ACK

RTPAudio

3PCC- Flow I

A BController

1. INVITE no SDP

2. 200 offer1

3. INVITE offer1

4. 200 OK answer1

5. ACK

6. ACK answer1

RTP

Property of Flow I

Advantage The call flow is easy

Disadvantage Retransmission time out

Suitable for the callee is a server The response time of server is short

3PCC- Flow IV

A BController

1. INVITE offer1 no m

2. 200 answer1 no m

4. INVITE no SDP

5. 200 OK offer2

8. ACK answer2

9. ACK

RTP

3. ACK

6. INVITE offer2’

7. 200 answer2’

Property of Flow IV

Advantage No retransmission timeout problem

Disadvantage Call flow is complicated

Suitable for the callee is a human

3pcc for session mobility

A BController C

1. INVITE offer1 no m

2. 200 answer1 no m 3. ACK

4. INVITE no SDP

5. 200 OK offer2

6. INVITE offer2’

7. 200 answer2’

8. ACK answer2

9. ACK

RTP9. BYE

10. ACK

Issues

Schulzrinne, H.; Xiaotao Wu; Sidiroglou, S.; Berger, S.; “ Ubiquitous computing in home networks”, Communications Magazine, IEEE , Volume: 41 , Issue: 11 , Nov. 2003 Pages:128 - 135

Issues

Seamless handover Handoff delay

The advantages of session mobility using SIP Flexible (Application layer) Easy to spread Extra application

Split session

Issues (cond.)

Change the call flow of 3PCC for suited the session mobility

Another method for session mobility using SIP？ REFER Method 3PCC

Split Session

Assume devices are SIP-enabled Third-party call control

INVITE speakerphonem=audioc=pc42

INVITE displaym=videoc=pc42

192.0.2.1

192.0.2.7

INVITE pc42m=videoc=192.0.2.7m=audioc=192.0.2.1

pc42

Category of spilt session

Split all session Split partial session Add new sessions Combine session

1. Split all session

video

microphone

speaker

VideoMicrophone

speaker

All media stream move from the origin device to other devices

A B

2. Split partial session

video

Microphonespeaker

VideoMicrophone

speaker

Some media stream keep on the original device and some media stream move to the other devices

A B

3. Add new sessions

video

microphone

speaker

Microphonespeaker

Additional media stream join the communication

A B

4. Combine session

VideoMicrophone

speaker

video

microphone

speaker

We communicate with many devices and want to continue communicate when changing device

A B

Basic REFER call flow

OriginatorA1

Final RecipientA2

RecipientB

INVITE/200/Ack

INVITE

100 Trying

ACK

RTP

202 Accepted

REFER

NOTIFY :100 Trying

200 OK

RTP

200 OKNOTIFY :200 OK

200 OK

Content type:message/sigfrag; version=2.0Body:SIP/2.0 100 Trying

No Body

BYE

200 OK

3pcc for session mobility

A BController C

1. INVITE offer1 no m

2. 200 answer1 no m 3. ACK

4. INVITE no SDP

5. 200 OK offer2

6. INVITE offer2’

7. 200 answer2’

8. ACK answer2

9. ACK

RTP9. BYE

10. ACK

Compare REFER and 3pcc

REFER Advantage

Popular SIP extension Disadvantage

Not suite for all category of split session

3pcc Advantage

More flexibility than REFER Additional application

Disadvantage Need additional device (Controller=B2BUA)

On-line Split session using REFER OriginatorA@ccu

Final RecipientA_video@ccu

RecipientB@nctu

INVITE (video)

100 Trying (video)

ACK (video)

202 Accepted (video)

REFER (video@ccu)

NOTIFY :100 Trying (v)

200 OK (v)

RTP

200 OK (video)

NOTIFY :200 OK (v)

200 OK (v)

Final RecipientA_audio@ccu

REFER (audio@ccu)

202 Accepted (audio) INVITE (audio)

100 Trying (audio)NOTIFY :100 Trying (a)

200 OK (a)200 OK (audio)

NOTIFY :200 OK (a)

200 OK (a)ACK (audio)

RTP

Add body to describe media type

Problem of Split partial sessionOriginator

A@ccuFinal RecipientA_video@ccu

RecipientB@nctu

INVITE

ACK

RTP

202 Accepted

REFER

RTP

200 OKNOTIFY :200 OK

200 OK

BYE

Video

Video / Audio

Video / Audio200 OK

Terminate all session with the

same call ID

re-INVITE

200 OK

ACK

RTPAudio

On-line split session using 3pcc

OriginatorA@ccu

Final RecipientA_video@ccu

RecipientB@nctu

Final RecipientA_audio@ccu

Controller

3pcc Flow IV

3pcc Flow IV

BYE

200 OK

RTP

RTP

RTPVideo / Audio

Split partial session using 3pcc

Controller

RTP

ACK

RTPVideo / Audio

3pcc Flow IV

Video

re-INVITE

200 OK

RTPAudio

OriginatorA@ccu

RecipientB@nctu

Final RecipientA_video@ccu

Add new sessions using 3pcc

OriginatorA@ccu

Final RecipientA_video@ccu

RecipientB@nctu

Final RecipientA_audio@ccu

Controller

3pcc Flow IV

3pcc Flow IV

BYE

200 OK

RTP

RTP

RTPVideo / Audio

Audio

Video

Nested REFER

RFC 3892 September 2004

Retrieval Session

Partial Session

Partial session One or more media streams of a

session are on a user agent The others are on the different

user agents.

Rules

Session Manager Has partial sessions Has all the information of the partial session, such as

configuration, security, and authorization All the partial session can be transferred, retrieved, and

terminated by the session manager Session User

Has partial sessions Only can transfer itself partial session to the session manager.

Free Node Keeps no any partial session

Session Manager Transformation

Session Manager Transformation (SMT) Used to support the session manager state transformation between

different user agents. Provide a session manager to transfer all the session information to a

session user or a free node. After SMT, the session user will become the session manager, and the

original session manager will become a session user or a free node.

The Flowchart of SMT at Session Manager

The Flowchart of SMT at Session User or Free Node

Session Manager Acquirement Provide a session user or a free node to

acquire all the session information from the session manager. Send the acquiring message from a session user

or a free node to a session manager Authentication information will be added into the

acquiring message. The session manager accepts the acquiring

message, all the session information will return to the session user or the free node

Session Manager Acquirement

Reference

Session Initiation Protocol (RFC 3261) Session Initiation Protocol Basic Call Flow Examples (RFC 3665) The SIP INFO Method (RFC 2976) The Session Initiation Protocol Refer Method (RFC 3515)