Introduction to VoIP

1. Introduction to VoIP • Benefits • Steps • Components • Protocols and standards • Issues

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)

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

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

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.

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.

8. Protocols and Standards

9. VoIP Environment (H.323 adopted) Gateway Router MCU H.323 gatekeeper Telephone H.323 terminal Packet switched network Circuit switched network

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

11. Establishing a Connection with H.323

12. Elements in an SIP Environment Redirect server • Proxy: forwards the communications to the user agent. If two organisations link then the proxy can be configured for the communications automatically • Registrar: This server records your location and how to contact you Location server User Agent Server (UAS) Internet User Agent Client (UAC) Local proxy server Remote proxy server • Redirect Server: This server will redirect communications to your current location if you have moved. • User 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

13. proxy Establishing a Connection with SIP Source: Juniper Networks

14. Establishing a Connection with SIP

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 • SIP Server: • OnDo SIP server • Router with build-in VoIP features • IP soft phones:Windows • Messenger 5.0 or higher • SJPhone • X-Lite

16. H323 Vs SIP

17. Key Issues • Quality of Service (QoS) • Scalability • Interoperability • Security • Emergency services • Integration with PSTN

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

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

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

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

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

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

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

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

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).

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)

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

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 • 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/

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

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

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/