Voice over IP VoIP

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Voice over IP - VoIP

• SMD151
• Peter.Parnes_at_ltu.se

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Overview

• What is VoIP?
• Toll ByPass
• RTP
• Control Protocols
• MGCP
• H323
• SIP in detail

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What is Voice over IP

• The packetisation and transport of classic public
  switched telephone system audio over an IP
  network.
• The analog audio stream is encoding in a digital
  format, with possible compression, and
  encapsulating it in IP for transport over your
  LAN/WAN or the public internet

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Categories of VoIP

• Private and Desktop Skype, Microsoft Netmeeting,
  ohphone, gphone, Asterisk, Marratech, etc.
• Enterprise Small IP phone deployments, IP PBX,
  Cisco Callmanager.
• ISP/Carrier Toll Bypass services, VOIP
  wholesale, using equipment by vendors such as
  Cisco, Lucent, Avaya, etc.
• Telco Grade Local service, Last mile delivery,
  total phone services, high dependability and
  availability.

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Toll ByPass
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How does it work?

• VoIP is not a protocol.
• VoIP is a collection of protocols and devices
  that allow for the encoding, transport and
  routing of audio calls over IP networks.
• PSTN ? VoIP ? PSTN
• Native VoIP ? PSTN
• Native VoIP ? Native VoIP

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The Guts

• RTP (Real-Time Transport Protocol)
• RTCP (Real-Time Control Protocol)
• RTP is a UDP stream with no intelligence for QoS
  or resource reservation
• Contains a packet number for detection of packet
  loss and re-sequencing of out of order packets.
• Unidirectional two streams in any call

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RTP

• RTP data packet header

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RTCP Sender Report

• Real-time Transfer Control Protocol Sender Report

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RTCP Receiver Report

• Receiver report

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Control Protocols - SIP

• SIP Session Initiation Protocol.
• SIP is a textual based client/server protocol.
  This makes debugging easy as its invite etc
  messages are human readable.
• INVITE sip021326001_at_203.109.130.945060userph
  one SIP/2.0
• Via SIP/2.0/UDP 10.30.60.205060
• From "Mr Anderson" ltsip099639951_at_10.30.60.205
  060gttag1ACC.8D00
• To ltsip021326001_at_203.109.130.945060gttag3598
  70F0-16CD
• Call-ID 00C0.95C9.5818.4013.1ACC.8D00_at_10.30.60.
  20
• CSeq 5257 ACK
• Content-Length 0
• Max-Forwards 70
• Contact sip099639951_at_10.30.60.205060

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Control Protocols MGCP

• MGCP Media Gateway Control Protocol.
• MGCP is a master/slave protocol where all the
  smarts resides in the gateway controller and not
  the gateway.

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Control Protocols H323
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Control Protocols H323

• value H323_UserInformation
• h323-uu-pdu
• h323-message-body releaseComplete
• protocolIdentifier 0 0 8 2250 0 4
• callIdentifier
• guid '3E940C894E1311D8A33E919F0987C365
  'H
• h245Tunneling TRUE

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H323
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H323 Call Setup Version 1

• H.323 call setup consisted of
• Q.931 call signaling (1st TCP connection)
• H.245 capability exchange (2nd TCP connection)
• H.225.0 establish logical channels (3rd TCP
  connection)
• Now send RTP...
• Each TCP connection requires a SYN exchange
  before data can be sent, plus the usual TCP
  timeouts on packet loss.
• Connection setup was excessively slow.

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H323 Evolution

• H.323 has evolved so that many of the original
  defects have been reduced.
• Unfortunately you still need to implement the old
  mechanisms too to be compliant.
• Two new methods of setup
• H.245 tunneling over the H.225.0 connection.
• H.323v2 FastStart

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H323 FastStart
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Control Protocols

• SIP
• MGCP
• H323
• Skinny Ciscos IP phone control protocol

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Summary

• What is VoIP?
• Toll ByPass
• RTP
• Control Protocols
• H323
• MGCP
• SIP

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Questions?

• ?

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SIP Overview

• SIP - Introduction
• SIP architecture and philosophy
• Methods Used in SIP
• SIP messages responses
• Security
• Summary
• References
• Nishita Vora Pandya

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SIP is

• An Application-layer control (signaling) protocol
  
• creating, modifying and terminating sessions with
  one or more participants.
• Sessions include
• Internet multimedia conferences,
• Internet telephone calls
• multimedia distribution.
• Members in a session can communicate via
• Multicast
• mesh of unicast relations
• combination of these.
• Text based
• Model similar to HTTP uses client-server model

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SIP Basic Functionality

• Supports 5 facets of communication
• User location determination of the end system to
  be used for communication
• User capabilities determination of the media and
  media parameters to be used
• User availability determination of the
  willingness of the called party to engage in
  communications
• Call setup "ringing", establishment of call
  parameters at both called and calling party
• Call handling including transfer and termination
  of calls.

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SIP Functionality (cont.)

• SIP can also initiate multi-party calls using a
  multipoint control unit (MCU) or fully-meshed
  interconnection instead of multicast.
• You can try in Marratech
• Internet telephony gateways that connect Public
  Switched Telephone Network (PSTN) parties can
  also use SIP to set up calls between them.

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Development of SIP

• SIP developed by Handley, Schulzrinne, Schooler,
  and Rosenberg
• - Submitted as Internet-Draft 7/97
• Assigned RFC 2543 in 3/99
• Goals Re-use of Maximum Interoperability with
  existing protocols
• Alternative to ITUs H.323
• - H.323 used for IP Telephony since 1994
• - Problems No new services, addressing, features
• - Concerns scalability, extensibility

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SIP Philosophy

• Internet Standard
• - IETF - http//www.ietf.org
• Reuse Internet addressing (URLs, DNS, proxies)
• - Utilizes rich Internet feature set
• Reuse HTTP coding
• - Text based
• Makes no assumptions about underlying protocol
• TCP, UDP, X.25, frame, ATM, etc.
• Support of multicast

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SIP Architecture

• SIP uses client/server architecture
• Client - originates message
• Server - responds to or forwards message
• Elements
• SIP User Agents (SIP Phones)
• SIP Servers (Proxy or Redirect - used to locate
  SIP users or to forward messages.)
• Can be stateless or stateful
• SIP Gateways
• To PSTN for telephony interworking
• To H.323 for IP Telephony interworking

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SIP Entities User Agents

• User Agents
• User Agent Client (UAC)
• Initiates SIP requests
• User Agent Server (UAS)
• Returns SIP responses

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SIP Entities Network Servers

• Network Servers
• Proxy Decides next hop and forwards request,
  relays call signaling , operates in a
  transactional manner, saves no session state
• Redirect Sends address of next hop back to
  client, redirects callers to other servers
• Registrar Accepts REGISTER requests from
  clients, maintains users whereabouts at a
  location server
• Diff. types may be co-located.

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SIP Operation

• SIP Addressing
• Locating a SIP Server
• Sending SIP Requests SIP Transactions
• SIP Methods
• SIP Responses
• Subsequent Requests and Responses

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Step 1SIP Addressing

• Uses Internet URLs
• Uniform Resource Identifiers
• Supports both Internet and PSTN addresses
• General form is name_at_domain
• To complete a call, needs to be resolved down to
  User_at_Host
• Examples
• sippeter_at_parnes.com
• sipJ.T. Kirk ltkirk_at_starfleet.govgt
• sip46-8-50555699_at_digisip.netuserphone
• sipguest_at_10.64.1.1

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Step 2 Locating a SIP server

• A caller first locates the appropriate server
• When client wants to send a request URI
• client will either send it to
• - Locally configured Proxy server
• - IP address port corresponding to the
• request URI similar to the one in step 1
• Client must determine IP address, port of server
  and the protocol to be used.

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Locating (cont.)

• Client
• Send the request to the servers IP address if
  the host part of request URI is an IP address
  otherwise
• Should try to contact a server at the port listed
  in request URI. If no port specified then try
  port 5060
• Use specified protocol if applicable
• otherwise use UDP if supported,
• if UDP fails or otherwise use TCP
• Find one or more address of server by querying
  DNS,
• NAPTR, SRV, CNAME, A
• Results MAY be cached.
• Capability to interpret ICMP messages must exist

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Step 3Send a SIP request

• Once the host part has been resolved to a SIP
  server, - client sends 1 / more SIP requests to
  that server
• receives 1 / more responses from the server.
• SIP Request-line (Messages) defined as
• ltMethodgt ltSPgt Request-URI ltSPgtSIP-Version
  ltCRLFgt
• (SPSpace, CRLFCarriage Return and Line
  Feed)
• (Method INVITE ACK OPTIONS
• BYE CANCEL
  REGISTER)
• Example
• INVITE sippeter_at_parnes.com SIP/2.0

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Order of Operation

• Step 1 Caller Issues Initial INVITE Request
• Step 2 Callee Issues Response
• Step 3 Caller Receives Response to Initial
• request
• Step 4 Caller or Callee Generate Subsequent
• requests
• Step 5 Receive Subsequent Requests
• Step 6 BYE to end session
• Step x CANCEL may be issued

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Simple Example
Audio, video, ...
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Methods (cont.)

• INVITE Initiates sessions
• - Session description included in message body
• Re-INVITEs used to change session state
• ACK confirms session establishment, can only be
  used with INVITE
• BYE terminates a session (hanging up)
• CANCEL cancels a pending invite
• REGISTER binds a permanent address to current
  location, may convey user data
• OPTIONS capability inquiry

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Redirect Example
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Redirect Example
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Relaying a Call
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Relay Example
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Relay and Topology Hiding
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SIP Negotiation

• Optimistic Call

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SIP Negotiation

• Pessimistic Call

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Multi-register Example
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Proxy Server Example
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Redirect Server Example
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SIP Responses

• SIP Responses defined as (HTTP-style)
• SIP-Version SP Status-Code SP Reason-Phrase
  CRLF
• (SPSpace, CRLFCarriage Return and Line Feed)
• Example
• SIP/2.0 404 Not Found
• First digit gives Class of response

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SIP Responses (cont.)

• 1xy Informational
• request received , continuing to process
  request
• 2xy Success
• action successfully recvd., understood
  accepted
• 3xy Redirection
• Further action to be taken to complete the
  request
• 4xy Client error
• request contains syntax error or cant be
  completed at this server
• 5xy Server error
• server fails to fulfill an apparently valid
  request
• 6xy global failure,
• request is invalid at any server

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Step 4 Generate Subsequent Requests

• Once the call has been established
• either the caller or callee may generate INVITE
  or BYE requests to change or terminate the call.
  
• For the desired call leg the headers are set as
  follows (both including any tags)
• - the To header field is set to the remote
  address, and
• - the From header field is set to the local
  address.
• The Contact header field may be different than
  the Contact header field sent in a previous
  response or request.
• The Request-URI may be set to the value of the
  Contact header field received in a previous
  request or response from the remote party, or to
  the value of the remote address. supports
  mobility

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SIP Requests Example

• Required Headers (fields)
• Via Shows route taken by request.
• Call-ID unique identifier generated by client.
• CSeq Command Sequence number
• generated by client
• Incremented for each successive request

INVITE sippicard_at_wcom.com SIP/2.0 Via
SIP/2.0/UDP host.wcom.com5060 From Alan
Johnston ltsipalan.johnston_at_wcom.comgt To Jean
Luc Picard ltsippicard_at_wcom.comgt Call-ID
314159_at_host.wcom.com CSeq 1 INVITE
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Via Field in Header

• The Request headers include a Via field
• The Via field indicates the path taken by the
  request so far.
• Every proxy adds a Via Header with its address to
  make sure that responses within a transaction
  take the same path (to avoid loops, or to make
  sure that same firewall will be hit on the way
  back)
• This prevents request looping and ensures replies
  take the same path as the requests, which assists
  in firewall traversal and other unusual routing
  situations.

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Via Headers and Routing

• Via headers are used for routing SIP messages
• Requests
• Request initiator puts address in Via header
• Servers check Via with senders address, then add
  own address, then forward. (if different, add
  received parameter)
• Responses
• Response initiator copies request Via headers.
• Servers check Via with own address, then forward
  to next Via address
• All Via headers are copied from request to
  response in order
• Response is sent to address in top Via header

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Via Header (cont.)
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VIA Example
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Via Example Response
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Step 5 Receiving Subsequent Requests 

• Subsequent to receipt, the following checks are
  made 
• 1. If the Call-ID is new,
• - the request is for a new call, regardless
  of the values
• of the To and From header fields.
•  2. If the Call-ID exists,
• - the request is for an existing call.
• - If the To, From, Call-ID, and CSeq values
  exactly match (including tags) those of any
  requests received previously,
• - the request is a retransmission. 
  
• 3. If there was no match to the previous step,
• - To From fields compared against existing
  call leg local and remote addresses.
• - If there is a match, the CSeq in the
  request gt last CSeq
• received on that leg,
• - the request is a new transaction for an
  existing call leg.

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Reliability

• If UDP is used
• -SIP client should retransmit a BYE, CANCEL,
  OPTIONS, or REGISTER request, exponential
  backoff, starting at a T1 second interval,
  doubling the interval for each packet, and
  capping off at a T2 second interval.
• -Retransmit a INVITE request with an interval
  that starts at T1 seconds, exponential back off,
  cease retransmissions if a provisional or
  definitive response recvd., or once it has sent a
  total of 7 request packets
• Clients using TCP do not need to retransmit
  requests

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Authentication Encryption

• SIP supports a variety of approaches
• end to end encryption
• hop by hop encryption
• Proxies can require authentication
• Responds to INVITEs with 407 Proxy-Authentication
  Required
• Client INVITEs again with Proxy-Authorization
  header.
• Stacked!!
• SIP Users can require authentication
• Responds to INVITEs with 401 Unauthorized
• Client INVITEs with Authorization header

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Features and Benefits
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Features and Benefits(cont.)
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SIP vs. H323

• H.323 uses TCP for setup
• UDP gives better timing control
• UDP is critical for very large proxies that wish
  to be stateless.
• H.323 uses ASN.1 Packed Encoding Rules
• For a long time only one company sold an ASN.1
  compiler that could compile H.323's ASN.1
• SIP uses an HTTP-like syntax.
• Much easier to write and to debug.
• Much experience with HTTP

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SIP Summary

• SIP is
• Relatively easy to implement
• Gaining vendor and carrier acceptance
• Very flexible in service creation
• Extensible and scaleable
• Appearing in products right now
• SIP provides its own reliability mechanism is
  therefore independent of the packet layer and
  only requires an unreliable datagram service

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But

• SIP is not
• Going to make PSTN inter-working easy
• Going to solve all IP Telephony issues (QoS)
• Secure !?
• SIP is still evolving and being extended as
  technology matures and SIP products are
  socialized in the marketplace.
• Newer RFCs 3261 through 3265

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Questions?

• ?