1. CSE5803 Advanced Internet Protocols and Applications (14) 1 14.1 Introduction Developed in recent years, for low cost phone calls (long distance in particular). Facilitate the convergence of voice and data services. Cost reduction achieved by compression coding, and efficient use of bandwidth through multiplexing. Voice is a real-time interactive service. The guarantee of Quality of Service (QoS) should be considered at the same time as improving bandwidth efficiency (although QoS suffers for many reasons). VoIP will not replace PSTN in the short term. The current aim is to provide a sound, lower cost alternative. Both ITU and IETF have proposed umbrella standards for VoIP.

2. CSE5803 Advanced Internet Protocols and Applications (14) 2 14.2 ITU-T H.323 VoIP Structure H.323 defines the procedure for multimedia communications over packet-switched networks. (The following diagram and explanation are from http://www.protocols.com)

3. CSE5803 Advanced Internet Protocols and Applications (14) 3 Gateway: Provides interoperability between different networks, converts signalling and media e.g., IP/PSTN signalling, audio codecs for voice compression coding. Gatekeeper: Manages a zone (collection of H.323 devices). –Required Functionality: Address translation, admissions control. –Optional Functionality: Call authorization, bandwidth management, supplementary services, directory services, call management services. H.323 Terminal: Endpoint on a LAN. Supports real-time, 2-way communications with another H.323 entity. Must support voice (audio codecs) and signalling (Q.931, H.245, Registration Admission Status -RAS). Optionally supports video and data e.g., PC phone or videophone, Ethernet phone.

4. CSE5803 Advanced Internet Protocols and Applications (14) 4 MCU: Multipoint Control Unit. Supports conferences between 3 or more endpoints. Contains multi-point controller (MC) for signalling. May contain multi-point processor (MP) for media stream processing. Can be stand-alone (i.e., PC) or integrated into a gateway, gatekeeper or terminal. H323 protocol stacks (www.protocols.com):

5. CSE5803 Advanced Internet Protocols and Applications (14) 5 14.3 IETF SIP VoIP Structure Session Initiation Protocol (SIP/RFC2543) is proposed by IETF for multimedia session initiation. (The following diagram and explanation are from http://www.protocols.com)http://www.protocols.com UAC (user agent client): Caller application that initiates and sends SIP requests. UAS (user agent server): Receives and responds to SIP requests on behalf of clients; accepts, redirects or refuses calls.

6. CSE5803 Advanced Internet Protocols and Applications (14) 6 SIP Terminal: Supports real-time, 2-way communication with another SIP entity. Supports both signalling and media, similar to H.323 terminal. Contains UAC. Proxy: Contacts one or more clients or next-hop servers and passes the call requests further. Contains UAC and UAS. Redirect Server: Accepts SIP requests, maps the address into zero or more new addresses and returns those addresses to the client. Does not initiate SIP requests or accept calls. Location Server: Provides information about a caller's possible locations to redirect and find proxy servers. May be co-located with a SIP server. SIP protocol stack:

7. CSE5803 Advanced Internet Protocols and Applications (14) 7 or TCP

8. CSE5803 Advanced Internet Protocols and Applications (14) 8 14.4 Voice and Audio/Video Encoding Standards Voice coding standards are provided by ITU-T for conversational voice compression coding. Sophisticated algorithms achieve better efficiency but cause longer coding delay. Faster hardware may speed it up a little. The following table is from http://www.protocols.comhttp://www.protocols.com Audio/Video encoding: MPEG-2 for high resolution and quality. MPEG-4 for low resolution and small screen applications.

9. CSE5803 Advanced Internet Protocols and Applications (14) 9 14.5 Functions of Real-time Transport Protocol (RTP) RTP (RFC1889) and RTCP (control protocol RFC1889) provides facilities for real time applications such as voice and video to be transported over internet. RTP is carried by UDP/IP and addresses only for the packetization of real time payloads at end systems. It is an end-to-end protocol that does not provide resource reservation or quality of service guarantee. There is no error recovery, either. The main functions of RTP/RTCP are timestamping, sequencing, framing, source identification, session control, and synchronisation. 14.6 Quality of Service (QoS) Issues The indicators of QoS are essentially delay, delay jitter (variation) and loss ratio. For real-time services, a higher layer protocol may not be able to compensate lower layer QoS degradation. Delay for voice service include coding, line propagation and queueing. Queueing delay is variable depending on routes. Congestion will cause loss (which can be replaced by silence).

10. CSE5803 Advanced Internet Protocols and Applications (14) 10 H323 or SIP does not have inherent mechanisms for QoS guarantee. IETF has proposed two basic mechanisms for QoS guarantee. The first model is Integrated Services (IntServ) based on Resource Reservation Protocol (RSVP), and the second is Differentiated Services (DiffServ). RSVP/IntServ acts on each link flow (end to end) and is therefore resource intensive. It is not scalable with existing hardware. The basic idea of DiffServ is similar to ATM (layer 2) service classes. It seeks to aggregate traffic flows with similar QoS requirements and handles them hop by hop at routers. The service class needs to be labeled with each IP datagram.