Vo ip sip

© 2007 Cisco Systems, Inc. All rights reserved. Cisco ConfidentialPresentation_ID 1
VoIP & SIP Signaling
Hussam El Kebbi

Presentation_ID 2© 2007 Cisco Systems, Inc. All rights reserved. Cisco Confidential
Overview
 VoIP Architecture
 What is VoIP ?
 Benefits of VoIP
 Components of a VoIP Network
 Quiz
 VoIP Telephone Call
 Overview on a VoIP Connection
 Analog/Digital Interfaces
 Steps of Conversion
 Transport Layer
 QoS in VoIP/Solutions for QoS Issues
 Calculating Bandwidth Requirements for VoIP
 Cisco IOS Configurations for VoIP
 Quiz

 SIP Architecture
 What is SIP?
 SIP Capabilities
 SIP URI / Components
 Quiz
 SIP Message Format
 SIP Message
 Request/Response Fields
 Header Fields
 Quiz
 SIP Call Flow
 Using Proxy Server/Using Multiple Servers
 How are the Codecs Negotiated / SDP Information
 SIP Security Mechanisms
 QUIZ
Overview

VoIP Architecture

VoIP Architecture
Voice over IP (VoIP) defines a way to carry voice calls over
an IP network including the digitization and packetization of
the voice streams
What is VoIP ?
http://www.cisco.com/en/US/tech/tk652/tk701/tsd_technology_support_protocol_home.html

Benefits of VoIP
 More efficient use of bandwidth and equipment
 Reduce operating costs
 Consolidated network expenses
 Improved employee productivity
 Access to new communication devices
VoIP Architecture
CCNP: Optimizing Converged Networks v5.0NT, Chapter 2

Components of a VoIP Network
These are the most common elements in VoIP networks:
VoIP Architecture

Quiz
 Define VoIP, and list two of its benefits ?
 List three of VoIP network components ?
VoIP Architecture

VoIP Telephone Call

To setup a VoIP communication we need:
 Convert analog voice to digital signals (bits)
 Now the bits have to be compressed in a good format for
transmission
 Insert our voice packets in data packets using a real-time protocol
(typically RTP over UDP over IP)
 We need a signaling protocol to call users (SIP - H.323)
 At Receiving we have to disassemble packets, extract data, then
convert them to analog voice signals
 All that must be done in a real time fashion cause we cannot waiting
for too long for a vocal answer! (QoS )
Overview on a VoIP connection
http://tldp.org/HOWTO/VoIP-HOWTO-4.html
VoIP Telephone Call

 Foreign Exchange Station (FXS)
 Foreign Exchange Office (FXO)
 Ear and Mouth (E&M)
Legacy Analog Interfaces in VoIP Networks
Digital Interfaces
Analog Interfaces
VoIP Telephone Call
Click me

Converting Analog Signals to Digital Signals
 Sample the analog signal (Sampling)
 Quantize sample into a binary expression (Quantization)
 Compress the samples to reduce bandwidth
Converting Digital Signals to Analog Signals
 Decompress the samples
 Decode the samples into voltage amplitudes
 Reconstruct the analog signal
VoIP Telephone Call
Steps of Conversion
Click me

To calculate the total bandwidth, find the total packet size, including all
the headers plus payload and divide by the payload size. Multiply the
result by the nominal bandwidth for the codec. The result is the total
bandwidth requirement.
VoIP Telephone Call
Calculating Bandwidth Requirements for VoIP
You can calculate the bandwidth using the Voice Codec Bandwidth Calculator at
http://tools.cisco.com/Support/VBC/do/
CodecCalc1.do.

Real-time applications such as voice and video require a guaranteed
connection with consistent and predictable delay characteristics.
IP does not guarantee reliability, flow control, error detection, or error
correction
 TCP offers both connection-oriented and reliable transmission
- Handles sequencing and error detection to ensure that the destination
application receives a reliable stream of data
 UDP, like IP, is a connectionless protocol.
- Routes data to its correct destination port but does not attempt to
perform any sequencing or to ensure data reliability
Transport Layer
VoIP Telephone Call

 RTP solves the problem enabling the receiver to put the packets back
into the correct order and not wait too long for packets that have either
lost their way or are taking too long to arrive
- RTP transports the digitized samples of real-time information
- The packets can be correctly reordered
- The packets can have appropriate delays inserted between packets
Transport Layer
VoIP Telephone Call
VoIP doesn't use TCP because it is too heavy for real time
applications

 Latency: Delay for packet delivery
 Jitter: Variations in delay of packet delivery
 Packet loss: Too much traffic in the network causes the network to
drop packets
 Burstiness of Loss and Jitter: Loss and Discards (due to jitter)
tend to occur in bursts
QoS (Quality of Service) is a major issue in VOIP
implementations, things to consider are:
VoIP Telephone Call
QoS in VoIP
http://www.voip-info.org/wiki/view/QoS

Solution for QoS issues
 Resource reservation : Make sure that the VoIP call has the
bandwidth needed allocated from point to point before the
conversation takes place.
 Prioritization: Here, the end point suggest a priority on the packets
and each router decides if it will honour this request or not.
 Network Traffic Tuning: Boxes you can add to a network to
manage bandwidth usage and create QOS even if the other network
devices don't support it.
VoIP Telephone Call
http://www.voip-info.org/wiki/view/QoS

Cisco IOS Configurations for VoIP
http://www.cisco.com/en/US/products/hw/routers/ps221/products_configuration_guide_chapter09186a008007c9bc.html

Quiz
 Which three components should be taken into
consideration when calculating the voice bandwidth needed
to set up a call on a VoIP network? (Choose three)
1. Voice payload size
2. RTP, UDP, and IP headers
3. Layer 2 encapsulation
4. Low latency queuing (LLQ)
5. Classification and marking of the voice traffic
6. Call Admission Control enabled on the network
VoIP Telephone Call

 Does VoIP use TCP as transport protocol?Why?
VoIP Telephone Call
Quiz

SIP Architecture

SIP Architecture
 The Internet Engineering Task Force's (IETF's)
standard for multimedia conferencing over IP
 A signaling protocol used to create, manage and
terminate sessions in an IP based network.
 A client/server protocol, which is similar to HTTP
 Influencing the marketplace, a growing number of IP
Telephony Service Providers (ITSP)/ cellular phone
providers, Microsoft real-time communication platforms,
and Cisco applications are based on SIP
Session Initiation Protocol (SIP) is :
What is SIP?

Presence, Instant Messaging and Voice

 Determine the location of the target end point
 Determine the media capabilities of the target end
point—Via Session Description Protocol (SDP)
 Determine the availability of the target end point
Establish a session between the originating and target
end point
 Handle the transfer and termination of calls
SIP Capabilities :
SIP Architecture
Click me

 A user of an online service
 An appearance on a multiline phone
 A mailbox on a messaging system
 A telephone number at a gateway service
SIP URIs have a format based on e-mail address formats, namely
user@domain. There are two common schemes. An ordinary SIP URI is of
the form: sip:bob@biloxi.com
The URI may also include a password, port number, and related parameters.
SIP Architecture
SIP Universal Resource Indicators
http://www.cisco.com/web/about/ac123/ac147/archived_issues/ipj_6-1/sip.html
A resource within a SIP configuration is identified by a URI.
Examples of communications resources include the following:

 User agent client (UAC)
 User agent server (UAS)
SIP Components
Functional Components :
 Proxy server : Perform call routing, authentication, authorization,
address resolution, and loop detection
 Redirect server : UAs and proxy servers can contact a redirect server to
find the location of an end point
 Registrar : Processes requests from UACs for registration of their current
location
SIP Architecture
SIP is a peer-to-peer protocol, can function in one of the
following roles:
Click me
SIP User
Agents

Quiz
 What is SIP?
 Name 3 Funtional Components of SIP Architecture?
SIP Architecture

SIP Message Format

SIP Message Format
SIP is a simple, ASCII text-based protocol that uses requests and
responses to establish communication among the various components in
the network
generic-message = start-line ;start-line = Request-Line / Status-Line
*message-header
CRLF ; carriage-return line-feed sequence
[ message-body ]
SIP Message
http://www.tech-invite.com/Ti-sip-abnf.html

Status-Line = SIP-Version SP Status code SP Reason-phrase CRLF
 SIP version : The SIP version being used.
 Status-code : A 3-digit integer result code of the attempt to understand and
satisfy the request.
 Reason-phrase : A textual description of the status code.
SIP Message Format
Response Fields
Request Fields
Request-Line = Method SP Request-URI SP SIP-Version CRLF
 Method : Register, Invite, Ack, Cancel, Bye, and Options
 Request-URI : It indicates the user or service to which this request is being
addressed
 SIP version : Is case-insensitive but implementations must send upper case

 Provisional (1xx): The request was received and is being
processed.
 Success (2xx): The action was successfully received,
understood, and accepted.
 Redirection (3xx): Further action needs to be taken in order to
complete the request.
 Client Error (4xx): The request contains bad syntax or cannot
be fulfilled at this server.
 Server Error (5xx): The server failed to fulfill an apparently valid
request.
 Global Failure (6xx): The request cannot be fulfilled at any
server.
SIP Message Format
Response
The SIP response types defined in RFC 3261 are in the
following categories:
www.cisco.com/web/about/ac123/ac147/archived_issues/ipj_6-1/sip.html

Header Fields
 To : Specifies the desired "logical" recipient of the request
 From : Indicates the logical identity of the initiator of the request
 CSeq : Provide a means to uniquely identify transactions
 Call-ID : Acts as a unique identifier to group together a series of messages
 Contact : Provides a URI whose meaning depends on the type of request or
response it is in.
 Via : Indicates the path taken by the request so far and indicates the path
that should be followed in routing responses
SIP Message Format
A valid SIP request formulated by a UAC MUST, at a
minimum, contain following header fields:
http://www.networksorcery.com/enp/protocol/sip.htm

INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 12.26.17.91:5060
Max-Forwards: 70
To: Bob <sip:bob@biloxi.com
From: Alice <sip:alice@atlanta.com;tag=1928301774
Call-ID: a84b4c76e66710@12.26.17.91
CSeq: 314159 INVITE
Contact: <sip:alice@atlanta.com>
Content-Type: application/sdp
Content-Length: 142
SIP Message Format
Request Example

Response Example
SIP/2.0 200 OK
Via: SIP/2.0/UDP server10.biloxi.com
Via: SIP/2.0/UDP bigbox3.site3.atlanta.com
Via: SIP/2.0/UDP 12.26.17.91:5060
To: Bob <sip:bob@biloxi.com;tag=a6c85cf
From: Alice <sip:alice@atlanta.com;tag=1928301774
Call-ID: a84b4c76e66710@12.26.17.91
CSeq: 314159 INVITE
Contact: <sip:bob@biloxi.com>
Content-Type: application/sdp
Content-Length: 131
SIP Message Format

Quiz
 SIP Message code basis are ?
 Binary
 ASCII
 What are the Fields of SIP Message ?
SIP Message Format

SIP Call Flow

SIP Call Flow
Figure 1, Using proxy server

Figure 2, Using multiple servers
SIP Call Flow

How are the Codecs negotiated?
 SDP is the protocol used by the UAs to tell each other what codecs they
support. SDP is embedded into the SIP Messages.
SDP is intended for describing multimedia sessions for the purposes
of session announcement, session invitation, and other forms of
multimedia session initiation
 SDP, defined in RFC 2327, describes the content of sessions, including
telephony, Internet radio, and multimedia applications
SIP Call Flow
Session Descripton Protocol (SDP)
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.6.2.10
s=SIP Call c=IN IP4 10.6.2.10
t=0 0
m=audio 24580 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/800
http://www.ietf.org/rfc/rfc2327.txt

 Media streams: A session can include multiple streams of differing
content. SDP currently defines audio, video, data, control, and
application as stream types.
 Addresses: SDP indicates the destination addresses, which may be a
multicast address, for a media stream.
 Ports: For each stream, the UDP port numbers for sending and
receiving are specified.
 Payload types: For each media stream type in use (for example,
telephony), the payload type indicates the media formats that can be
used during the session.
 Start and stop times: These apply to broadcast sessions, for example,
a television or radio program. The start, stop, and repeat times of the
session are indicated.
 Originator: For broadcast sessions, the originator is specified, with
contact information. This may be useful if a receiver encounters technical
difficulties.
SIP Call Flow
SDP Information :

 Why SDP is used ?
SIP Call Flow
Quiz

SIP Security
 Internet Telephony uses a variety of signaling protocols,
such as H.323, SIP, MGCP and MEGACO, for initiating
VOIP calls.
 SIP, like other Internet Protocols, is vulnerable to known
Internet attacks.
 VOIP suffers from all known attacks associated with
any Internet application or subsystem
SIP Security Mechanisms
Saverio Niccolini, Ph. D.Research Staff Member @ Network Laboratories NEC Europe Ltd

SIP Client-A drops the call just initiated
DoS Attack

Call Hijack
Threats

Identity Theft

Registration and call signaling/media should be
authenticated
 End-to-end
- Digest authentication (challenge - response)
- S/MIME
 Hop-by-hop
- TLS, IPsec
- SIPS

Ways to ensure signaling security in SIP:
 HTTP Digest: prone to eavesdropping, replay, and MiTM attacks.
Provides authentication only.
 TLS: Hop-by-hop SIP transport security; not end-to-end! Provides
confidentiality, authentication, encryption.
 S/MIME : End-to-end signaling and body security. Provides
confidentiality, authentication, encryption.
 IPSec: Layer 3 security. Provides confidentiality and encryption.

Quiz
 Which are Hop-by-hop / End-to-end SIP transport
security?
 TLS
 IPSEC
 HTTP Digest
 S/MIMe End-to-end

Presence, Instant Messaging and Voice
Thank You
Hussam El Kebbi

SIP Components / Servers / Services
SIP User
Agents
Registrar Redirect
Location
Database
SIP Proxy
SIP Servers /
Services
REGISTER
“Here I am”
INVITE
“I want to talk
to another UA”
Proxied INVITE
“I’ll handle it for
you”
“Where is this
name/phone#?”
3xx Redirection
“They moved,
try this address”
SIP User
Agents
SIP-GW
http://www.cisco.com/

Vo ip sip

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