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VoIP Basics


   Rajib Deka
   Sr. Programmer
   Siemens Ltd.
   Chennai-100.


                    Rajib Deka   1
Agenda
   Internet Basics
   Protocol Layering
   Voice Over IP
   VoIP Architecture
   VoIP Network
   VoIP Protocols
   SIP Basics
   Conclusion

                Rajib Deka   2
Internet Basics
   The basic building block of networks is the IP datagram.

   Analogy to datagram - a postcard with
     Destination address

     Return address

     A small amount of text



   A postcard might inform you of a friend’s holiday travels or
    remind you of a dentist’s appointment.

   The postal service doesn’t care which application (friend or
    dentist) sent the postcard—it just carries processed wood pulp
    with black marks.



                        Rajib Deka                    3
Protocol Layering




            Rajib Deka   4
Voice Over IP - Introduction
   A recent application of Internet technology – Voice
    over IP (VoIP): Transmission of voice over Internet.

   How VoIP works
       Continuously sample audio
       Convert each sample to digital form
       Send digitized stream across Internet in packets
       Convert the stream back to analog for playback

   Why VoIP
       IP telephony is economic; High costs for traditional
        telephone switching equipments.
       Call setup: call establishment, call termination, etc.
       Backward compatibility with existing PSTN (Public Switched
        Telephone Network)
                        Rajib Deka                  5
Voice Over IP - Introduction
   IP Telephony Standards:
       ITU (International Telecommunication Union) controls
        telephony standards.
       IETF (Internet Engineering Task Force) controls TCP/IP
        standards.
       Audio is encoded using a well-known standard such as
        Pulse Code Modulation (PCM).

   UDP is used for transport:
       lower overhead: audio must be played as it arrives.
       Playback cannot be stopped to wait for a retransmitted
        packet.


                        Rajib Deka                 6
VoIP Architecture
   VoIP Server:
       Media Gateway.
       Media Gateway Controller.
       Signaling Gateway.
       IP PBX and Proxy.

   VoIP Client:
       Soft Phones.
       IP Phones.

                       Rajib Deka   7
VoIP Network




          Rajib Deka   8
VoIP Network




          Rajib Deka   9
VoIP Protocols
   Main complexity of VoIP: Call setup and call management.

   The process of establishing and terminating a call is called
    Signaling.

   In traditional telephone system, signaling protocol is SS7.

   In VoIP, signaling protocols are:
       SIP (Session Initiation Protocol), by IETF
       H.323, by ITU
       Megaco & MGCP, jointly by IETF and IUT.

   Audio Signaling:
       RTP: Real-time Transport Protocol.
       RTCP: Real Time Control Protocol.

   VoIP signaling protocols should be able to interact with SS7.

                           Rajib Deka                   10
SIP Basics
   SIP: Session Initiation Protocol. Invented by IETF.
   SIP defines three main elements that comprise a
    signaling system:
       User Agent: IP phone or applications
       Location servers: stores information about user’s location
        or IP address
       Support servers:
           Proxy Server: forwards requests from user agents to another
            location.
           Redirect Server: provides an alternate called party’s location
            for the user agent to contact.
           Registrar Server: receives user’s registration requests and
            updates the database that location server consults.


                          Rajib Deka                      11
SIP Characteristics
   Operates at the application layer.

   Encompasses all aspects of signaling, e.g. location of called
    party, ringing a phone, accepting a call, and terminating a call.

   Provides services such as call forwarding.

   Relies on multicast for conference calls.

   Allows two sides to negotiate capabilities and choose the media
    and parameters to be used.

   SIP URI is similar to email address. (with prefix “sip:”) E.g.
    sip:rajib.deka@siemens.com


                         Rajib Deka                     12
SIP Methods
   Six basic message types, known as methods:




                Rajib Deka         13
SIP Session
                          User agent A contacts DNS
                           server to map domain name in
                           SIP request to IP address.

                          User agent A sends a INVITE
                           message to proxy server that
                           uses location server to find the
                           location of user agent B.

                          Call is established between A
                           and B. Then media session
                           begins.

                          Finally, B terminates the call by
                           sending a BYE request.



              Rajib Deka                     14
RTP and RTCP
   RTP used to send real-time streams of data across a network is
    simply called the Real Time Protocol (RTP for short). RTP has
    been originally defined by IETF.

   RTCP accompanies RTP and is used to transmit control
    information about the RTP session. RTCP packets are send only
    from time to time since there is a recommendation that the RTCP
    traffic should consume less than 5 percent of the session
    bandwidth. The most important content types carried in RTCP
    packets include:
         Information about call participants (for example, name and e-mail
          address)
         Statistics about the quality of the transmission (for example inter-
          arrival jitter and the number of lost packets).

   RTCP to monitor Quality of Service (QoS).



                           Rajib Deka                         15
VoIP and QoS
   QoS (Quality of Service) is a major issue in VOIP
    implementations. The issue is how to guarantee that
    packet traffic for a voice or other media connection
    will not be delayed or dropped due interference from
    other lower priority traffic.
   Things to consider are
       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.



                        Rajib Deka                16
VoIP Developer’s choice
   Language:
       C, C++, Java for Core protocol stack development.
       Java, C# for middle tier or application development.

   Open Source VoIP Servers (Linux Based)
       Asterisk PBX (Multi protocol support)
       OpenSIPS (for SIP only)
       sipXecs (for SIP only)

   VoIP Clients
       X-lite.
       VoIP Communicator.

   Open Source SIP stack
       JainSIP
       sipXecs
       PJSIP



                             Rajib Deka                        17
Conclusion
   IP telephony or VoIP refers to the transmission of voice
    telephone calls over IP networks.

   Hot area both in research and market because of low cost

   Challenge in backward compatibility with PSTN

   The complexity of IP telephony is on signaling. Both ITU and
    IETF propose signaling standards.

       H.323, by IUT
       SIP, by IETF, offering similar functions to H.323, but simpler than
        H.323.
       Both are competing to be recognized as #1 signaling protocol.




                          Rajib Deka                     18
Rajib Deka   19
Rajib Deka   20

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Vo Ip Rajibdeka

  • 1. VoIP Basics Rajib Deka Sr. Programmer Siemens Ltd. Chennai-100. Rajib Deka 1
  • 2. Agenda  Internet Basics  Protocol Layering  Voice Over IP  VoIP Architecture  VoIP Network  VoIP Protocols  SIP Basics  Conclusion Rajib Deka 2
  • 3. Internet Basics  The basic building block of networks is the IP datagram.  Analogy to datagram - a postcard with  Destination address  Return address  A small amount of text  A postcard might inform you of a friend’s holiday travels or remind you of a dentist’s appointment.  The postal service doesn’t care which application (friend or dentist) sent the postcard—it just carries processed wood pulp with black marks. Rajib Deka 3
  • 4. Protocol Layering Rajib Deka 4
  • 5. Voice Over IP - Introduction  A recent application of Internet technology – Voice over IP (VoIP): Transmission of voice over Internet.  How VoIP works  Continuously sample audio  Convert each sample to digital form  Send digitized stream across Internet in packets  Convert the stream back to analog for playback  Why VoIP  IP telephony is economic; High costs for traditional telephone switching equipments.  Call setup: call establishment, call termination, etc.  Backward compatibility with existing PSTN (Public Switched Telephone Network) Rajib Deka 5
  • 6. Voice Over IP - Introduction  IP Telephony Standards:  ITU (International Telecommunication Union) controls telephony standards.  IETF (Internet Engineering Task Force) controls TCP/IP standards.  Audio is encoded using a well-known standard such as Pulse Code Modulation (PCM).  UDP is used for transport:  lower overhead: audio must be played as it arrives.  Playback cannot be stopped to wait for a retransmitted packet. Rajib Deka 6
  • 7. VoIP Architecture  VoIP Server:  Media Gateway.  Media Gateway Controller.  Signaling Gateway.  IP PBX and Proxy.  VoIP Client:  Soft Phones.  IP Phones. Rajib Deka 7
  • 8. VoIP Network Rajib Deka 8
  • 9. VoIP Network Rajib Deka 9
  • 10. VoIP Protocols  Main complexity of VoIP: Call setup and call management.  The process of establishing and terminating a call is called Signaling.  In traditional telephone system, signaling protocol is SS7.  In VoIP, signaling protocols are:  SIP (Session Initiation Protocol), by IETF  H.323, by ITU  Megaco & MGCP, jointly by IETF and IUT.  Audio Signaling:  RTP: Real-time Transport Protocol.  RTCP: Real Time Control Protocol.  VoIP signaling protocols should be able to interact with SS7. Rajib Deka 10
  • 11. SIP Basics  SIP: Session Initiation Protocol. Invented by IETF.  SIP defines three main elements that comprise a signaling system:  User Agent: IP phone or applications  Location servers: stores information about user’s location or IP address  Support servers:  Proxy Server: forwards requests from user agents to another location.  Redirect Server: provides an alternate called party’s location for the user agent to contact.  Registrar Server: receives user’s registration requests and updates the database that location server consults. Rajib Deka 11
  • 12. SIP Characteristics  Operates at the application layer.  Encompasses all aspects of signaling, e.g. location of called party, ringing a phone, accepting a call, and terminating a call.  Provides services such as call forwarding.  Relies on multicast for conference calls.  Allows two sides to negotiate capabilities and choose the media and parameters to be used.  SIP URI is similar to email address. (with prefix “sip:”) E.g. sip:rajib.deka@siemens.com Rajib Deka 12
  • 13. SIP Methods  Six basic message types, known as methods: Rajib Deka 13
  • 14. SIP Session  User agent A contacts DNS server to map domain name in SIP request to IP address.  User agent A sends a INVITE message to proxy server that uses location server to find the location of user agent B.  Call is established between A and B. Then media session begins.  Finally, B terminates the call by sending a BYE request. Rajib Deka 14
  • 15. RTP and RTCP  RTP used to send real-time streams of data across a network is simply called the Real Time Protocol (RTP for short). RTP has been originally defined by IETF.  RTCP accompanies RTP and is used to transmit control information about the RTP session. RTCP packets are send only from time to time since there is a recommendation that the RTCP traffic should consume less than 5 percent of the session bandwidth. The most important content types carried in RTCP packets include:  Information about call participants (for example, name and e-mail address)  Statistics about the quality of the transmission (for example inter- arrival jitter and the number of lost packets).  RTCP to monitor Quality of Service (QoS). Rajib Deka 15
  • 16. VoIP and QoS  QoS (Quality of Service) is a major issue in VOIP implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic.  Things to consider are  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. Rajib Deka 16
  • 17. VoIP Developer’s choice  Language:  C, C++, Java for Core protocol stack development.  Java, C# for middle tier or application development.  Open Source VoIP Servers (Linux Based)  Asterisk PBX (Multi protocol support)  OpenSIPS (for SIP only)  sipXecs (for SIP only)  VoIP Clients  X-lite.  VoIP Communicator.  Open Source SIP stack  JainSIP  sipXecs  PJSIP Rajib Deka 17
  • 18. Conclusion  IP telephony or VoIP refers to the transmission of voice telephone calls over IP networks.  Hot area both in research and market because of low cost  Challenge in backward compatibility with PSTN  The complexity of IP telephony is on signaling. Both ITU and IETF propose signaling standards.  H.323, by IUT  SIP, by IETF, offering similar functions to H.323, but simpler than H.323.  Both are competing to be recognized as #1 signaling protocol. Rajib Deka 18