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
„ At session or media level

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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
„ At session or media level
„ a=(attributes)
„ Describe additional attributes

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 Ordering of Fields
„ Session Level „ Media level
„ Protocol version (v) „ Media description (m)
„ Origin (o) „ Media info (i)
„ Session name (s) „ Connection info (c)
„ Session information (i) „ Optional if specified at the
session level
„ URI (u)
„ Bandwidth info (b)
„ E-mail address (e)
„ Encryption key (k)
„ Phone number (p)
„ Attributes (a)
„ Connection info (c)
„ Bandwidth info (b)
„ Time description (t)
„ Repeat info (r)
„ Time zone adjustments (z)
„ 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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 OPTIONS Method

„ 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

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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
 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
Example of e2e Resource Reservation [1/2]

„ 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
„ 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]

„ 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
„ 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

„ Each participant deals with

network access permission at
its own end.
 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
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.
PSTN Interworking
„ 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.
 Interworking with H.323

„ SIP-H.323 interworking gateway

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 Summary

„ The future for signaling in VoIP networks

„ Simple, yet flexible
„ Easier to implement
„ Fit well with the media gateway control protocols
„ SIP is the protocol of choice for the evolution
of third-generation wireless networks.
„ SIP-based mobile devices will become available
„ SIP-based network elements will be introduced
within mobile networks.

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