The Session Description Protocol

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1 Internet Telephony

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)

SIP uses SDP in an answer/offer mode.

An agreement between the two parties as to the types of media they are willing to share

RFC 3264 (An Offer/Answer Model with SDP)

To describe how SDP and SIP should be used together

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The Structure of SDP

SDP simply provides a format for describing session information to potential session

participants.

Text-based Protocol

The Structure of SDP

Session Level Info

Name of the session

Originator of the session

Time that the session is to be active

Media Level Info

Media type

Port number

Transport protocol

Media format

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SDP Syntax

A number of lines of text

In each line

field=value

field is exactly one character (case-significant)

Session-level fields

Media-level fields

Begin with media description field (m=)

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Mandatory Fields

v=(protocol version)

o=(session origin or creator)

s=(session name), a text string

For multicast conference

t=(time of the session), the start time and stop time

For pre-arranged multicast conference

m=(media)

Media type

The transport port

The transport protocol

The media format, an RTP payload format

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Optional Fields [1/3]

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

i=(session information)

A text description

At both session and media levels

It would be somewhat superfluous, since SIP already supports the Subject header.

u=(URI of description)

Where further session information can be obtained

Only at session level

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Optional Fields [2/3]

e=(e-mail address)

Who is responsible for the session

Only at the session level

p=(phone number)

Only at the session level

c=(connection information)

Network type, address type and connection address

At session or media level

b=(bandwidth information)

In kilobits per second

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Optional Fields [3/3]

r=(repeat times)

For regularly scheduled session a session is to be repeated

How often and how many times

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 purposes of encrypting and decrypting the media

a=(attributes)

Describe additional attributes

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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 the session level

Bandwidth info (b)

Encryption key (k)

Attributes (a)

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Subfields [1/3]

Field = <value of subfield1> <value of subfield2>

<value of subfield3>.

Origin

Username, the originator’s login id or “-”

session ID

A unique ID

Make use of NTP timestamp

version, a version number for this particular session

network type

A text string

IN refers to Internet

address type

IP4, IP6

address, a fully-qualified domain name or the IP address

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Subfields [2/3]

Connection Data

The network and address at which media data will be received

Network type

Address type

Connection address

Media Information

Media type

Audio, video, data, or control

Port

Format

List the various types of media format that can be supported

According to the RTP audio/video profile

m= audio 45678 RTP/AVP 15 3 0

G.728, GSM, G.711

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Subfields [3/3]

Attributes

To enable additional information to be included

Property attribute

a=sendonly

a=recvonly

value attribute

a=orient:landscape used in a shared whiteboard session

rtpmap attribute

The use of dynamic payload type

a=rtpmap:<payload type> <encoding name>/<clock rate>

[/<encoding parameters>].

m=video 54678 RTP/AVP 98

a=rtpmap 98 L16/16000/2

16-bit linear encoded stereo (2 channels) audio sampled at 16kHz

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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 the sessions.

The entity headers identifies the message body.

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SIP Inclusion in SIP Messages

Fig 5-15

G.728 is selected

INVITE with multiple media streams

Unsupported should also be returned with a port number of zero

An alternative

INVITE

m=audio 4444 RTP/AVP 2 4 15 a=rtpmap 2 G726-32/8000 a=rtpmap 4 G723/8000 a=rtpmap 15 G728/8000

CONNECT

m=audio 6666 RTP/AVP 15 a=rtpmap 15 G728/8000

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SIP and SDP Offer/Answer Model

Re-INVITE is issued when the server replies with more than one codec.

With the same dialog identifier (To and From headers, including tag values), Call-ID and Request-URI

The session version is increased by 1 in o= line of message body.

A mismatch

488 or 606

Not Acceptable

A Warning header with warning code 304 (media type not available) or 305 (incompatible media type)

Then the caller issues a new INVITE request.

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Determine the capabilities of a potential called party

Accept Header

Indicate the type of information that the sender hopes to receive

Allow Header

Indicate the SIP methods that Boss can handle

Supported Header

Indicate the SIP extensions that can be supported

OPTIONS Method

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SIP Extensions and Enhancements

RFC 2543, March 1999

SIP has attracted enormous interest.

Traditional telecommunications companies, cable TV providers and ISP

A large number of extensions to SIP have been proposed.

SIP will be enhanced considerably before it

becomes an Internet standard.

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183 Session Progress

It has been included within the revised SIP spec.

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)

For SIP-PSTN-SIP connections

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The Supported Header

The Require Header

In request

A client indicates that a server must support certain extension.

In response

421, extension required

The Supported header

RFC 2543 – Require: header (client -> server)

420 (bad extension) – server -> client

Can be included in both requests and responses

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SIP 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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SIP Event Notification

Several SIP-based

applications have been

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

E.g., Instant messaging

RFC 3265 has addressed the issue of event notification.

SUBSCRIBE and NOTIFY

The Event header

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SIP for 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.

A MESSAGE request does not establish a SIP dialog.

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SIP REFER Method

To enable the sender of the request to instruct the receiver to 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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Reliability of Provisional Responses

Provisional Responses

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

Are not answered with an ACK

If the messages is sent over UDP

Unreliable

Lost provisional response may cause problems when interoperating with other network

180, 183 → Q931 alerting or ISUP ACM

To drive a state machine

E.g., a call to an unassigned number

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

“The number you have called has been changed”

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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RFC 3262

Reliability of Provisional Responses in SIP

Supported: 100rel

RSeq Header

Response Seq

+1, when retxm

RAck Header

Response ACK

In PRACK

RSeq+CSeq

PRACK

Prov. Resp. ACK

Should not

Apply to 100

Default timer value = 0.5 s

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The 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

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

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

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

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 (Chapter 8)

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 type of usage.

Policing functions at the edge of the network

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

Reserving network

resources in advance of altering the called user

A new draft –

“Integration of Resource Management and SIP”

By using the provisional responses and UPDATE method

By involving extensions to SDP

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Example of e2e Resource Reservation [1/2]

SDP for initial INVITE

v=0o=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

SDP for 183 response

v=0o=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]

SDP for UPDATE

v=0o=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

SDP for 200 response

v=0o=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

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Example of Aggregate- based Reservation

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

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Usage of SIP for Features/Services

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

To carry MIME content as well as an SDP description

SIP address is a URL

Click-to-call applications

The existing supplementary services in traditional telephony

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

Proxy invokes various types of advanced feature logic.

Policy server (call-routing, QoS)

Authentication server

Use the services of an IN SCP over INAP

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Call Forwarding

On busy

486, busy here

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

originally sent to User 2.

Contact: header in 200 response

Call-forwarding-on-no- answer

Timeout

CANCEL method

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Consultation Hold

A SIP UPDATE

User A asks User B a question, and User B

need to check with User C for the correct answer.

User B could use the REFER method to

transfer the call to User

C.

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PSTN Interworking

A SIP URL to a telephone number

A network gateway

PSTN – SIP – PSTN

MIME media types

For ISUP

SIP for Telephony (SIP-T)

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

SIP-H.323 interworking gateway

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