Voip basics


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Voip basics

  1. 1. VOIP - Basics<br />Vamsi Krishna, <br />Manager – Business Solutions<br />Works for Tata Communications<br />
  2. 2. VOIP – What is this all about<br />VOIP is a family of technologies that enables IP networks to be used for voice applications such as telephony, conferencing, messaging etc.<br />In the traditional PSTN world, end users need not be aware/worry on various elements involved in telephone calls.<br />How ever in VOIP awareness on Signalling, Bearer Channels, Codecs, Database services etc would be required.<br />
  3. 3. VOIP – What is this all about<br />SS7 – Signaling System 7 is the signaling type being used in PSTN – Out band Signaling i.e control messages are carried on dedicated channels.<br />In VOIP – signaling are of two types:<br />Peer to Peer (SIP & H.323) In this type the end points have intelligence to create, establish sessions directly, understand and interpret control messages.<br />Client Server (SCCP, MGCP, H.248) In this type, a Call Agent holds all the intelligence. A scenario could that when a phone gets off hook, media gateway doesn’t provide the dial tone but sends the event alert to “call agent” which instructs the media gateway to provide the dial tone to the phone.<br />
  4. 4. VOIP – What is this all about<br />G.711 – a Codec/ compression mechanism to digitize and packetize the analog voice calls to a 64 kbps stream. Bandwidth per voice call required would be approx 80 kbps.<br />G.729 – a Codec/compression mechanism to digitize and packetize the analog voice calls to a 8 Kbps stream. Bandwidth per voice call required would be approx 24-28 kbps.<br />
  5. 5. Components of VOIP<br />IP Phone/End Points – End user interface for telephone calls.<br />Gatekeeper: Provides Call Admission Control, Bandwidth Management, QOS, Network Translation.<br />Gateway: Does IP-TDM conversion.<br />Call Agent: Similar to Gatekeeper<br />MCU: Mixes multiple calls and supports conferencing<br />Application Servers: Supporting Messaging, Presence, Voice Mail, IVR etc.<br />
  6. 6. VOIP Signaling – H.323<br />H.323 is an ITU Standard.<br />It’s a Peer to Peer Signaling Protocol.<br />Its an umbrella of protocols defined to support synchronized voice, video and data.<br />Primarily built for Multimedia Conferencing.<br />This is a connection-less protocol. <br />This evolved from H.320 ISDN Standard.<br />
  7. 7. VOIP Signaling – MGCP<br />MGCP is a Client Server Signaling Protocol.<br />It’s a IETF Standard<br />Emerging standard for PSTN Gateway Control or Thin Device Control.<br />Useful since each end point need not deploy full signaling intelligence such as H.323<br />In this, the gateway/end point communicates/notifies events to a more intelligent central device called call agents which instructs the gateway/endpoint on further steps.<br />
  8. 8. VOIP Signaling – SIP<br />SIP is an IETF Standard, less complex than H.323.<br />Session Initiation Protocol.<br />It’s a Peer to Peer Signaling Protocol.<br />Its defines on the commands and responses to set up and close sessions between two end points.<br />It also details on Security, Proxy and transport protocol services.<br />SIP along with other partner protocols such as SDP (Session Discovery Protocol), SAP (Session Announcement Protocol) provides information on multicast sessions to the users.<br />SIP is a text bases protocol heavily borrowing functionalities from HTML such as request and respond model, similar header and response codes.<br />It uses URL Style addressing. <br />
  9. 9. VOIP Signaling – SCCP<br />Cisco Proprietary<br />Works between Cisco End Points and Call Manager.<br />
  10. 10. Comparing VOIP Signaling Protocols<br />H.323 calls for complex configuration between Gateways. This will require configuration of dial plans, route patters on the Gateways.<br />Ex can be H.323 between CUCM – Cisco 3845 connecting to PSTN World. Cisco 3845 is configured with full dial plans. Both Q931 and Q921 terminates on 3845.<br />In MGCP Scenario, Q921 terminates on to the router while Q931 is backhauled on to the CUCM. All Route plans, dial patterns are configured on the CUCM, <br />In SIP Scenario is very much came as H.323.<br />
  11. 11. VOIP Considerations<br />Jitter – Should be as low as possible<br />Latency – one way to be less than 150 ms<br />Low packet loss – as this is a UDP.<br />High availability<br />In TDM, dedicated bandwidth, dedicated timeslots, no variable delay or jitter.<br />PSTN provides 99.999<br />
  12. 12. More on VOIP<br />RTP – This is the protocol which carries the actual voice/video payload between the two points.<br />RTCP is a sister protocol providing call control/stats for the RTP Stream.<br />RTP & RTCP function over UDP.<br />RTP is generally between 16384 to 32677 port number. RTP is even port numbers, RTCP is odd port number.<br />cRTP & sRTP have been developed to enhance RTP.<br />RTP defines on how the payload is to be transported such as payload identification, time stamping, sequencing etc.<br />Using Time Stamps & Sequence Numbers on the Header the receiving end point can buffer and play back appropriately.<br />RTP doesn’t request for retransmission.<br />RTP Header is of 40 bytes Size.<br />
  13. 13. What is a PBX<br />A small exchange owned by company.<br />This lets the company take fewer number of trunks from telco. Typically number of lines required are 10% of the extensions.<br />