1. Introduction to VoIP Benefits Steps Components Protocols and standards Issues Version 2.1 2009Slide 1

2. Benefits of a VoIP Network More efficient use of bandwidth and equipment Lower transmission costs Consolidated network expenses Improved employee productivity through features provided by IP telephony: IP phones are complete business communication devices Directory lookups and database applications (XML) Integration of telephony into any business application Software-based and wireless phones offer mobility. Access to new communications devices (such as PDAs and cable set-top boxes) Version 2.1 2009Slide 2

3. VoIP Steps Convert the Analogue voice to digital data (ADC) Compress the digital ready for transport (CODEC) A number of different codec’s are used depending on the scenario G.711 G.722 G.723 List of which software supports which standard http://compare.ozvoip.com/codecsupport.php Place packets on the network for transport A signaling protocol to call users. E.g. H323 or SIP. Receiver converts packet back to analogue information to listen to Version 2.1 2009Slide 3

4. Basic Voice Encoding Step 1: Sample the analog signal. Step 2: Quantize sample into a binary expression. Step 3: Compress the samples to reduce bandwidth. Step 4: Decompress the samples. Step 5: Decode the samples into voltage amplitudes, rebuilding the PAM signal. Step 6: Reconstruct the analog signal from the PAM signals.

5. Components of a VoIP Network Version 2.1 2009Slide 5

6. Components of a VoIP Network Telephones provide telephony features to users. can be IP phones, software-based telephones operated on PCs, or traditional telephones (analog or ISDN). Gateways interconnect the VoIP network with traditional telephony devices. These can be analog or ISDN telephones, faxes, circuit-switched PBX systems, or public switched telephone network (PSTN) switches. Multipoint control units required for conferences. If more than two parties are involved in a call, all members of the conference send their media to the MCU. The MCU mixes the media and then sends the media to all participants. Application servers provide XML-based services to IP phones. IP phone users have access to directories and databases through XML applications. Version 2.1 2009 Slide 6

7. Components of a VoIP Network Gatekeepers provide Call Admission Control (CAC) to prevent the network from being over subscribed. As well, CAC translates telephone numbers or names to IP addresses for call routing in an H.323 network. Call agents provide call control, CAC, bandwidth control, and address translation services to IP phones or Media Gateway Control Protocol (MGCP) gateways. Video endpoints provide video telephony features to users. As with audio-only calls, video calls need a multipoint control unit for conferences. For videoconferences, the multipoint control unit has to be capable of mixing video and audio streams. Version 2.1 2009Slide 7

8. 8 Protocols and Standards Version 2.1 2009

9. 9 VoIP Environment (H.323 adopted) Gateway Telephone Router H.323 gatekeeper H.323 terminal MCU Packet switched networkCircuit switched network Version 2.1 2009

10. Setup Procedure of an H.323 Call Registration and admission Call setup Terminal capability negotiation,channel setup and master-slave detection Stable call established and proceeds Close channel Call teardown Disengagement RAS Q.931 H.245 RTP/RTCP H.245 Q.931 RAS Version 2.1 2009Slide 10

11. Establishing a Connection with H.323 Version 2.1 2009Slide 11

12. Elements in an SIP Environment Local proxy server Remote proxy server Redirect server Location server User Agent Server (UAS) User Agent Client (UAC) Internet lProxy: forwards the communications to the user agent. If two organisations link then the proxy can be configured for the communications automatically lRegistrar: This server records your location and how to contact you lRedirect Server: This server will redirect communications to your current location if you have moved. lUser Agent: This is the application which is requesting the service or the server which is responding. Communications are peer to peer. If connecting to the PSTN this would be a user agent Version 2.1 200912

13. Establishing a Connection with SIP lSource: Juniper lNetworks proxy Version 2.1 2009Slide 13

14. Establishing a Connection with SIP Version 2.1 2009Slide 14

15. SIP Open standard Text based protocol, similar to HTTP Support voice, video, chat, interactive game, virtual reality The SIP address is identified by a SIP URL: user@host. Examples of SIP URLs: sip:hostname@vovida.org sip:m28@192.168.0.5 sip:294284@www.staffs.ac.uk Useful SIP server and IP soft phone lSIP Server: lOnDo SIP server lRouter with build-in VoIP features lIP soft phones:Windows lMessenger 5.0 or higher lSJPhone lX-Lite Version 2.1 2009Slide 15

16. H323 Vs SIP SIPH.323 PHILOSOPHY "New World" - a relative of Internet protocols - simple, open and horizontal "Old World" - complex, deterministic and vertical IETF ITU Carrier-class solution addressing the wide area Borne of the LAN - focusing on enterprise conferencing priorities STATUS Industry endorsed Popularity due to the fact that it was the first set of agreed-upon standards Many vendors developing products The majority of existing IP telephony products rely on the H.323 suite Version 2.1 2009Slide 16

17. Key Issues Quality of Service (QoS) Scalability Interoperability Security Emergency services Integration with PSTN Version 2.1 2009Slide 17

18. Timely delivery Two factors will effect this The majority of business networks are Ethernet IPv4 version does not support any sort of Quality of service Both of the technologies used in the majority of networks are “Best Effort” This indicates that the network will do its best to deliver as quickly as possible It is NOT guaranteed This is a problem for phone conversations where delivery of data in a timely manner is essential VOIP must as far as possible at least provide the same service as the PSTN Version 2.1 2009Slide 18

19. Delays Official figures are given by the ITU-T These figures are intended as a guideline for usability of telecommunications including VOIP 0 to 150ms Acceptable for most user applications 150 to 400ms Acceptable delay for usage within international telecommunications > 400ms Unacceptable for general use. Although it is recognised that there will be situations where this guideline will be exceeded. Echo Version 2.1 2009Slide 19

20. Jitter Voice packets enter the network at a constant rate. Voice packets may arrive at the destination at a different rate or in the wrong order. Jitter occurs when packets arrive at varying rates. Jitter Since voice is dependent on timing and order, a process must exist so that delays and queuing issues can be fixed at the receiving end. The receiving router must: Ensure steady delivery (delay) Ensure that the packets are in the right order Version 2.1 2009Slide 20

21. Impact on existing LANs With proper planning there will not be a concern for VOIP But it does require consideration in the network plan significant bandwidth will be required and the larger the organisation the more network traffic will be saturated by voice calls Rather than just adding it to the current infrastructure, new and separate network can be allocated for the VOIP traffic This can be used as a backup anyway to the main network Version 2.1 2009Slide 21

22. Scalability Ability to add more telephony equipment as the company grows Network bandwidth and other issues may have an effect on scalability Version 2.1 2009Slide 22

23. Transporting voice data Commonly for data traffic IP is used in conjunction with TCP In VoIP networks though guaranteed delivery is not a requirement A few lost packets is fine, with the human ear being able to interpret missing information A paper by Intel discusses a maximum packet loss of 1% before the end user can hear the loss http://www.intel.com/network/csp/pdf/8539.pdf Uses TCP for call setup VoIP packets are transported with a combination of UDP and RTP Version 2.1 2009Slide 23

24. UDP User Datagram Protocol (UDP) Intended for speed rather than reliability Widely used were it is more important that the packets arrive quickly i.e. audio, video stream, online games Connectionless service Small header No support Packet ordering No support for packet delays On its own UDP would be useless for VoIP No packet ordering would cause voice signals to be corrupted Delays They can cause echoes in the conversation which would ruin the quality of the conversation Version 2.1 2009Slide 24

25. RTP Real Time Transport Protocol (RTP) The RTP is used to give some level of reliability to UDP Application layer protocol Allows for sequencing in a packet reordering Supports time stamping of the packets It does not give the full functionality of TCP as this is not required Widely used for streaming applications and Multimedia Version 2.1 2009Slide 25

26. Voice Encapsulation Digitized voice is encapsulated into RTP, UDP, and IP. By default, 20 ms of voice is packetized into a single IP packet.

27. Voice Encapsulation Overhead Voice is sent in small packets at high packet rates. IP, UDP, and RTP header overheads are enormous: For G.729, the headers are twice the size of the payload. For G.711, the headers are one-quarter the size of the payload. Bandwidth is 24 kbps for G.729 and 80 kbps for G.711, ignoring Layer 2 overhead.

28. RTP Header Compression Compresses the IP, UDP, and RTP headers Is configured on a link-by-link basis Reduces the size of the headers substantially (from 40 bytes to 2 or 4 bytes): 4 bytes if the UDP checksum is preserved 2 bytes if the UDP checksum is not sent Saves a considerable amount of bandwidth

29. When to Use RTP Header Compression Use cRTP: Only on slow links (less than 2 Mbps) If bandwidth needs to be conserved Consider the disadvantages of cRTP: Adds to processing overhead Introduces additional delays Tune cRTP—set the number of sessions to be compressed (default is 16). Version 2.1 2009Slide 29

30. Interoperability Interoperability between VoIP and PSTN, ISDN, GSM, GPRS Addressing schemes incompatible Charging users on the PSTN, etc., once off the internet segment Network architecture Gateway close to IP user (best quality) Gateway close to telephone user (lowest cost) Version 2.1 2009Slide 30

31. Slide 31 Connecting to PSTN Unless the devices can connect to the PSTN they will remain a fringe use The standards allow for normal type phones to be attached Businesses are now recognising the benefits Especially those who have invested heavily in the networks Members of the public are using this technology increasingly Cheap international calls Other service provided like, Video, text Version 2.1 2009

32. VoIP Regulations PSTN operators have a set of services they MUST provide due to regulation The consideration is should this be the same for VOIP ? Voice services are regulated IP services are not This is currently being investigated in the UK by OFCOM Findings from November 2004 Can be found here www.ofcom.org.uk/consult/condocs/new_voice/anew_voice/?a=87101 www.ofcom.org.uk/consult/condocs/new_voice/anew_voice/?a=87101 In November 2006 OFCOM agreed that VOIP numbers can have the prefix of 056, most operators though will still use geographic numbers such as 0161 = Manchester http://www.ofcom.org.uk/media/mofaq/telecoms/voip_faq/ Version 2.1 2009Slide 32

33. VoIP Regulations Initial considerations VOIP operators should not required to offer all of the services PSTN operators do Although desirable the 999/911/112 service does not have to be offered Although the customers must be informed of this This will also be reviewed as the market develops to possible make the operators support some sort of 999 service Version 2.1 2009Slide 33

34. VoIP and Emergency Services Why is a 999 call such a problem in VOIP ? In traditional the PSTN the current position of you phone is known You are physically connected to the network at that point Mobile phone operators Can find your location via the base station you are attached to Using multiple signals your exact location can be discovered This means if you phone 999 but you do not know your location the operator can find it for you In VOIP this is not correct, you are just a IP address You may be connecting into the exchange in London, but physically located in Manchester Version 2.1 2009Slide 34

35. Conclusion VoIP is a topical solution for number of reasons Many voice over IP options available 3% packets drop threshold No guesswork networks Security and reliability issues Big shows coming up VoIP for Business http://www.voipforbusiness.co.uk/http://www.voipforbusiness.co.uk/ Version 2.1 2009Slide 35

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