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  1. 1. [CN]Chapter 8 [CT]Voice over IP - H.232 and SIP Overview of VoIP......................................................................................................................................3 What Is VoIP?...........................................................................................................................................6 How Does VoIP Work?.........................................................................................................................6 Protocols Related to VoIP...................................................................................................................10 Signaling Protocols..............................................................................................................................10 VoIP Call-Control Protocols...............................................................................................................13 H.323.......................................................................................................................................................14 Terminals.............................................................................................................................................20 Gateway ..............................................................................................................................................21 Gatekeeper...........................................................................................................................................22 Multipoint Control Unit.......................................................................................................................23 H.323 Protocols...................................................................................................................................24 Audio Codec ...................................................................................................................................25 Video Codec ...................................................................................................................................26 H.225 RAS (Registration, Admission, and Status).........................................................................27 H.225 Call Signaling ......................................................................................................................27 H.245 Control signaling .................................................................................................................27 RTP (Real-Time Transport Protocol)..............................................................................................28 RTCP (Real-Time Transport Control Protocol) .............................................................................28 SIP (Session Initiation Protocol).........................................................................................................28 MGCP .................................................................................................................................................29 BICC (Bearer Independent Call Control) ...........................................................................................31 Routing Protocols....................................................................................................................................33 RIP (Routing Information Protocol) ...................................................................................................34 OSPF (Open Shortest Path First) Protocol .........................................................................................34 SPF Algorithm ....................................................................................................................................35 Border Gateway Protocol ...................................................................................................................36 Inter-autonomous system routing ...................................................................................................37 Intra-autonomous system routing ...................................................................................................38 Pass-through autonomous system routing ......................................................................................38 RSVP (Resource Reservation Protocol) .............................................................................................39 Transport Protocols.............................................................................................................................42 Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-1 5/6/2010
  2. 2. RTP (Real-Time Transport Protocol) .............................................................................................43 RTP Headers ...............................................................................................................................44 RTP Control Protocol (RTCP) ...................................................................................................45 IPv6......................................................................................................................................................45 SIP a Signaling Protocol..........................................................................................................................46 What Is SIP?........................................................................................................................................48 SIP Architecture..............................................................................................................................49 SIP Calls via a Proxy Server ..............................................................................................................54 Comparison of SIP to H.323...................................................................................................................56 Complexities of H.323 Versus SIP......................................................................................................56 Scalability............................................................................................................................................58 Extensibility.........................................................................................................................................62 If SIP is better, why is H.323 important? ...........................................................................................66 Figure 8-1 Comparing PSTN and VoIP voice call setup........................................................................11 Figure 8-2 Relationship between signaling and media flow...................................................................17 Figure 8-3 Components of a multimedia communication service...........................................................19 Figure 8-4 Gatekeeper.............................................................................................................................22 Figure 8-5 H.323 Zones...........................................................................................................................23 Figure 8-6 Interdomain routing...............................................................................................................36 Figure 8-7 Signaling separate from VoIP converstion............................................................................42 Table 8-1 Compression algorithms...........................................................................................................8 Table 8-2 H.323 Protocol Standards.......................................................................................................15 Table 8-3 ITD-T audio codec recommendations.....................................................................................25 Table 8-4 IP number codes and descriptions..........................................................................................51 Table 8-5 SIP and PSTN signal comparison...........................................................................................53 Table 8-6 Comparing SIP and H.323 features.........................................................................................61 Table 8-7 Comparing H.323 control service with SIP............................................................................64 [OBJ HD]After reading this chapter and completing the exercises, you will be able to: [BEG OBJ]Appreciate, define, and discuss VoIP technology and its time sensitivity when using IP as a transport mechinism. [BEG BL] Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-2 5/6/2010
  3. 3. • Understand that H.323 is a suite of protocols describing the rules for call processing and discuss the role of routers, gateways, and gatekeepers with regard to VoIP. • Discuss how voice is digitized, packetized, compressed, transmitted across an IP and PSTN network in nearly real time including the role of codecs in this process. • Understand the function of the latest VoIP protocol, SIP and how it is used to provide VoIP services. • Discuss and compare the strengths and weaknesses of the two major VoIP protocols H.323and SIP and compare their potential future with regard to audio and video transmission over an IP network. [END BL] [A HD]Overview of VoIP [BT A HD]IP telephony works in fundamentally different ways from the traditional telephone network that has carried voice communications for the past 100 years. Traditional voice is sent in a continuous stream over a circuit switched network. The longer the circuit the higher the tariff, or cost. Despite long silences during a conversation, the call accumulates a charge for every second the circuit is open. Even the migration from analog to digital circuits did little to change this model. But in recent years IP (Internet Protocol) has open the door for changing that tradition. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-3 5/6/2010
  4. 4. [BT]Just as data has been broken down into packets (small packages of signals), audio streams (waves) can be sampled using a DSP (digital signal processor), converted to binary 1s and 0s, packetized and sent over an IP-based network in data like packets. A packet switched IP network may be a local area network, a wide area network, a Telco’s core network or the Internet. Packet switching offers more efficient use of network resources. An IP packet is formed by a device’s network interface adapter that chops up a stream of signals into manageable chunks, adds source and destination addresses, formats them into a specific frame type, and presents them to the network. The packet is then routed along the most suitable route to its destination. [BT]The packets whether data, voice, or image may take different paths from node to node as they travel throughout a network. Taking different paths the packets will arrive at their destination out of order, because different paths vary in distance. They are then reassembled (put into order) and delivered in the form as they were initially entered into the network. Data packets will go to a data device and voice packets will go to a telephone. It is important to note that when packets arrive at their destination, they will arrive with a significant delay. This delay is due to the distance traveled and the sequencing required to put them in order at the destination. [BT]Errors in transmission for data are not a problem. If the data reaches the destination in error it can be resent. Errors in transmission and delay for voice are not acceptable. A caller's voice must reach its destination in real time. If voice packets are delayed more than 250 milliseconds from end to end, the sound quality will be unacceptable and customers will complain. Another voice packet concern is packet loss. What happens when one or more consecutive voice packets are lost? This will be answered later in this chapter. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-4 5/6/2010
  5. 5. [BT]To ensure voice quality, it is critical to evaluate what other demands are being placed on the LAN and WAN that may prevent voice traffic from getting to its destination within the time budget requirement (250 millisecond). Essentially, it needs to be prioritized over other types of traffic. Voice must be given the highest priority. There are many challenges ahead in migrating from the PSTN to an IP network: [BEG BL] • From a business perspective VoIP (Voice over Internet Protocol) means replacing presently existing systems like PBX and Centrex with IP-PBXs and IP-Centrex). • For the Telephone companies it means replacing their Class 5 switches with IP softswitches. • For residential subscribers it means access to a broadband network [END BL] [NOTE]In this chapter you will be introduced to numerous H.xxx protocols and G.xxx protocols. H.323, which will be the focus of the first half of this chapter, is analogous to TCP/IP. TCP/IP is not a single protocol but a protocol suite (many protocols), likewise H.323 is a suite of protocols. Be careful not to feel overwhelmed with so many new protocols. Tables are provided to assist you until they become familiar. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-5 5/6/2010
  6. 6. [BT]Internet telephony has now existed for about ten years but only recently are great strides being made in call quality to the point where companies like Skype and Vonage are providing telephone services utilizing the Internet to carry IP to IP calls and IP to PSTN calls and presently at a significant dollar savings. [A HD]What Is VoIP? [BT A HD]Softswitch is a product driven by a need to incorporate intelligence into VoIP (Voice over Internet Protocol) networks and interface IP networks with PSTN (Public Switched Telephone Network) as well as coordinate telephony features across networks. As discussed in the previous chapter, the first applications of softswitch were the gatekeepers (gateway controllers) that were incor- porated in VoIP networks. The importance of VoIP protocols relative to softswitch is that they are the building blocks that make VoIP possible. The popularity and applications of VoIP as seen by industry leaders will grow at a rapid rate from 2006 for the next five to seven years. With that in mind let us see how VoIP works and identify critical areas of concern. [B HD]How Does VoIP Work? Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-6 5/6/2010
  7. 7. [BT B HD]Softswitch is considered to be nearly synonymous with VoIP. However, it also works with TDM (Time Division Multiplexing) and ATM (Asynchronous Transfer Mode) networks. The first process in an IP voice application is the digitization of the speaker's voice. The next step is typically the suppression of unwanted signals and compression of the voice signal. Because IP is a protocol designed to interconnect networks of varying kinds and sizes, more processing is required than for smaller networks. The network addressing scheme can often be very complex, requiring a process of encapsulating (placing one packet inside another) and, as data moves along a network; repackaging, readdressing, and reassembling the data as it moves from one device to other devices. [BT]When each packet arrives at its destination, its sequence number is checked in order to place the packets in the proper-sequence. A decompression algorithm is used to restore the data to its original form, and clock synchronization and delay-handling techniques are used to ensure proper spacing of the packets. Since packets are transported through the network over different routes, they do not arrive at their destination in sequential order. To account for this, incoming packets are stored for a very short time in a jitter buffer to wait for late-arriving packets. The length of time that packets are held in the jitter buffer will vary depending on characteristics of the network. [BT]Due to the nature of IP networks, a number of the packets can be lost or possibly delayed, especially during periods of network congestion. Some packets are actually discarded due to errors that occurred during transmission. Damaged, lost, or delayed, packets result in deterioration of voice quality. For conventional error-correction techniques incoming blocks of data (not voice) containing Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-7 5/6/2010
  8. 8. errors are discarded, and the receiving device requests the retransmission of the packet; thus, the message that is finally delivered is exactly the same as the original message. Since VoIP is time- sensitive and cannot wait for retransmission, sophisticated error detection and correction procedures are used to fill in any gaps created by errors. This process uses stored portions of the incoming speaker's voice and then approximates the contents of the missing packets. Therefore, the voice sound heard by the receiver is not exactly the sound transmitted. [BT]The previous description details the movement of voice over an IP medium. The rest of the chapter will describe the building blocks and fundamentals of a VoIP network. You will require uninterrupted time while studying this material, since it contains many new term, acronyms, and concepts you have recently acquired from earlier chapters. Lets begin right after the voice has been digitized, and now is being prepared to be placed in packets. [BT]First, the system examines the recently digitized waveform to determine if it contains voice signals. Secondly, compression algorithms are employed to reduce the number of binary bits that must be sent to the called party. Codecs enable noise suppression and the compression of voice streams. Compression algorithms include G.723, G.728, G.729 and G.711, see table 8-1 to see the bandwidths per data packet of the G.xxx codecs. [TB HEAD]Table 8-1 Compression algorithms Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-8 5/6/2010
  9. 9. [TB TXT] Bandwidth Used per Bandwidth Used per Bandwidth Use for Call (includes IP Call (includes IP Type of Audio Data Packets headers) with 30 ms headers) with 20 ms Codec only data packets data packets G.711 64 kbps 80 kbps 88 kbps G.721 32 kbps 48 kbps 56 kbps G.723 6 kbps 22 kbps Not applicable G.728 16 kbps 32 kbps 40 kbps G.729a 8 kbps 24 kbps 32 kbps All H.323 compliant devices must support a minimum the G7.11 audio codec [END TBL] [BT]Following compression, voice signals are packetized and VoIP protocols added. Some storage of data occurs during the process of collecting voice data, since the transmitter must wait for a certain amount of voice signals to be collected before it can form a packet and be transmitted via the network. Information is added to the packet to facilitate its transmission across the network. Each packet will need to contain the address of its destination and a sequencing number in case the packets do not arrive in time. [NOTE]Due to the complexity of this new technology, and considering the amount of new terms and phraseology, you will encounter a few terms used that will not be elaborated on in the content of this chapter. These are terms that are better left to a more advanced VoIP courses. The terms are introduced to allow you to be aware that what is being covered is an abbreviation of what is ultimately available Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-9 5/6/2010
  10. 10. and to provide exposure regarding audio and video functionality. Cautions will be noted periodically. As always highlighted terms will appear in the Key words section of this chapter. [B HD]Protocols Related to VoIP [BT B HD]VoIP is built on a collection of the protocols. Those protocols are loosely analogous to PSTN and are broken down into three categories: access, switching, and transport. These three PSTN categories of protocols are similar to VoIP: signaling, routing, and transport. These three VoIP categories will now be discussed. [B HD]Signaling Protocols [BT B HD]The process of setting up a VoIP call is roughly similar to that of a PSTN circuit switched call. A media gateway must be configured with parameters to allow predefined use of telephony fea- tures. When the calling and called parties agree on how to communicate and the signaling criteria is established, the media stream over which the packetized voice conversation will flow is established. Signaling establishes the virtual circuit over the network for that media stream. Signaling is independent of the media flow. Today two types of protocols are currently popular for VoIP: H.323 and SIP. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-10 5/6/2010
  11. 11. [FIG]Figure 8-1 Comparing PSTN and VoIP voice call setup Media Gateway See figure 8-1 left side – Circuit switched system Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-11 5/6/2010
  12. 12. Phones are connected directly into the circuit switched system. [BEG BL] • Call signaling: The system detects a call request and extends the call to the destination. Negotiation of the type of connection usually occurs as part of the call signaling itself. • Media exchange: When the call is answered, the circuit-switched system must bridge the voice stream. Both call signaling and media exchanges are centralized. [END BL] See figure 8-1 right side – Media Gateway Phones connect to Call Manager through a network of routers. [BEG BL] • Call signaling: Media Gateway control detects a call request and extends the call to the destination. • Media control (sometimes, but not always, part of call signaling): When the destination answers, the endpoints must negotiate a codec and exchange addresses for purposes of exchanging media. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-12 5/6/2010
  13. 13. • Media exchange: The phones exchange media directly with each other. The media often follows a completely different set of routers from the call signaling. Call signaling and media control are centrally managed, but the high bandwidth media is distributed. [END BL] [B HD]VoIP Call-Control Protocols [BT B HD]The main VoIP call signaling and call-control protocols are: [BEG BL] • H.323 (a protocol for the transmission of real-time audio, video and data over packet switched networks). H.323 is the ITU-T recommendation with the largest installed bass, basically because it has been around the longest and no other protocol choices existed before H.323. • SGCP (Simple Gateway Control Protocol) was developed in 1998 to reduce the cost of endpoints (gateways) by having the intelligent call-control occur in a centralized platform (gateway controller). • IPDC (Internet Protocol Device Control) is very similar to SGCP, but it has many other mechanisms for OAM&P (operations, administration, management, and provisioning) than SGCP. OAM&P is crucial to carrier networks since it covers how equipment is maintained and deployed. • MGCP (Media Gateway Control Protocol) is basically SGCP with a few additions for OAM&P. Its origin was in late 1998. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-13 5/6/2010
  14. 14. • SIP (Session Initiation Protocol) is being developed as a media-based protocol that enables end devices (endpoints and gateways) to be more intelligent and enable enhanced services down at the call-control layer. [END BL] [BT]Only H.323 and SIP will be discussed at length in this book. MGCP will be given minimal coverage. These are presently the most widely used Call-Control protocols. SGCP and IPDC will not be explained further in this chapter. First H.323, the oldest and most distributed of the two, will be discussed. [A HD]H.323 [BT A HD]H.323 is the ITU-T (International Telecommunication Union-Telecommunications) Standardization Sector recommendation for packet based multimedia communication. Like TCP/IP, H.323 is comprised of a number of sub-protocols: [BEG BL] • H.225 is RAS (registration, admission, status) for call signaling, and control. H.323 also defines a set of call control, channel setup and codec specifications for transmitting real-time video and voice over networks that don't offer guaranteed service or QoS (quality of service) Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-14 5/6/2010
  15. 15. • As a transport, H.323 uses RTP (Real Time Protocol), an Internet IETF (Engineering Task Force) standard designed to handle the requirements of streaming real-time audio and video via the Internet. [END BL] See Table 8-2 for a summary of the H.323 protocol suite [TB HEAD]Table 8-2 H.323 Protocol Standards [TB TXT] Feature Protocol Call Signaling H.225 Media Control H.245 Audio Codecs G.711, G.722, G.723, G.728, G.729 Video Codecs H.261, H.263 Data Sharing T.120 Media Transport RTP/RTCP [END TBL] Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-15 5/6/2010
  16. 16. [BT]H.323 was the first VoIP protocol for interoperability among the early VoIP gateway/gatekeeper vendors. Unfortunately, the promise of inter operability between diverse vendors platforms did not materialize with the adoption of H.323. Figure 8-2 details the relationship between signaling and media flow. This relationship between transport and signaling is very similar to the PSTN in that SS7 (Signaling System 7) as that used in VoIP. In figure 8-2 IP phone 8000 is calling IP phone 8001. First the signaling process begins. [BEG BL] • Phone 8000 by taking the phone off hook which automatically signals the media gateway of its intent to place a call • The media gateway gives 8000 a dial tone • After phone 8000 dials the last digit of the called parties number, 8001 • The call manager checks to see if 8001 is on or off hook • If on hook the call manager gives each IP phone the IP address of the other • And the call manager has completed the signaling process • Now IP phone 8000 send a ringing signal to 8001 when 8001 goes off hook the conversation begins the media flow begins • When either IP phone 8000 or 8001 goes on hook the call is terminated. [END BL] Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-16 5/6/2010
  17. 17. [BT]Had the 8001 IP phone been off hook at the time 8000 called the media gateway would have sent a busy signal to 8000. [FIG]Figure 8-2 Relationship between signaling and media flow Media Gateway IP 8000 8001 IP Phone – IP Phone – Calling party Called party Call setup Signaling Media Flow [BT]H.323 specifies the components, protocols, and procedures providing multimedia communication over packet-based networks. Packet-based networks include IP-based or IPX (Internet packet Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-17 5/6/2010
  18. 18. exchange)-based LANs (local area networks), ENs (enterprise networks), MANs (metropolitan area networks), and WANs (wide area networks). H.323 can be applied in a variety of mechanisms: [BEG BL] • audio only (IP telephony) • audio and video (video telephony) • audio and data • and audio, video, and data • multipoint-multimedia communications [END BL] H.323 provides myriad services and can be applied in a wide variety of areas: [BEG BL] • Consumer • Business • Entertainment applications [END BL] Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-18 5/6/2010
  19. 19. [BT]Interworking with Other Multimedia Networks the H.323 standard specifies four kinds of components, when networked together; provide point-to-point and point-to-multipoint multimedia communication services: [BEG BL] • terminals • gateways • gatekeepers • MCUs (multipoint control units) [END BL] [FIG]Figure 8-3 Components of a multimedia communication service Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-19 5/6/2010
  20. 20. H.323 H.323 Terminal Terminal H.323 H.323 IP Network MCU Gateway H.323 H.323 Gatekeeper Gateway PSTN ISDN [B HD]Terminals [BT B HD]For the most part, terminals when mentioned in this chapter refer to telephones. However, an H.323 terminal used for real-time bidirectional multimedia communications can either be a stand- alone device running H.323 and multimedia applications or PC (personal computer). It supports audio communications and can support video or data communications. Because the basic service provided by an H.323 terminal is audio communications, an H.323 terminal plays a key role in IP-telephony ser- vices. The primary goal of H.323 is to interwork with multimedia terminals: Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-20 5/6/2010
  21. 21. [BEG BL] • H.310 terminals on Broadband-Integrated Services Digital Network (B-ISDN), • H.320 terminals on ISDN, • H.321 terminals on B-ISDN • H.322 terminals on guaranteed QoS LANs. • H.323 terminals may be used in multipoint conferences. • H.324 terminals on wireless networks [END BL] [BT]These H.323 protocols will be further described as you progress through the chapter. [B HD]Gateway [BT B HD]A gateway connects two dissimilar networks. An H.323 gateway provides connectivity between an H.323 network and a non-H.323 network. For example, a gateway can connect and provide communication between an H.323 terminal and a TDM network. This connectivity of dissimilar net- works is achieved by translating protocols for call setup and release, converting media formats between different networks, and transferring information between the networks connected by the gateway. A gateway is not required, however, for communication between two terminals (telephones) on the same H.323 network. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-21 5/6/2010
  22. 22. [B HD]Gatekeeper [BT B HD]It is the focal point for all calls within an H.323 network. Although they are not required, gatekeepers provide important services such as addressing, authorization, and authentication of terminals and gateways, bandwidth management, billing, and statistics. Gatekeepers may also provide call-routing services. Typically gatekeepers are used when more than one media gateway is in use. If there is no gatekeeper the media gateway performs the duties of the gatekeeper. In Figure 8-4 the gatekeeper interfaces with the two media gateways. The dashed lines are not direct line connections from the gatekeeper to the gateways but rather logical paths The physical path is into the IP network (the solid line) [FIG]Figure 8-4 Gatekeeper Gatekeeper Media Gateway Media Gateway IP IP IP IP WAN Voice Router Voice Router IP Phones IP Phones Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-22 5/6/2010
  23. 23. [B HD]Multipoint Control Unit [BT B HD]MCUs (Multipoint Control Units) provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU. The MCU manages conference resources, negotiates between terminals for the purpose of determining the audio or video codec (coder/decoder) to use, and may handle the media stream. The gatekeepers, gateways, and MCUs are logically separate components of the H.323 standard but can be implemented as a single physical device. [BT]H.323 Zone - An H.323 zone is a collection of all terminals, gateways, and MCUs managed by a single gatekeeper. A zone includes at least one terminal and may include gateways or MCUs. A zone has only one gatekeeper. See Figure 8-5. [FIG]Figure 8-5 H.323 Zones Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-23 5/6/2010
  24. 24. Media Gateway Gatekeeper IP IP IP WAN Router Router IP Phones IP Phones MCU [BT]A zone may be independent of network topology and may be comprised of multiple network segments that are connected using routers or other devices. [B HD]H.323 Protocols [BT B HD]Bear in mind that H.323 is independent of the packet network and the transport protocols over which it runs. A few important protocols specified by H.323 are listed and discussed in the following sections. They are: [BEG BL] • audio codecs • video codecs Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-24 5/6/2010
  25. 25. • H.225 RAS (registration, admission, and status) • H.225 call signaling • H.245 control signaling • RTP (Realtime Transport Protocol) • RTCP (Real Time Control Protocol) [END BL] [C HD]Audio Codec [BT C HD]An audio codec encodes the audio signal from the microphone for transmission on the transmitting H.323 terminal and an audio codec decodes at the destination translates on the receiving H.323 terminal. Because audio is the minimum service provided by the H.323 standard, all H.323 terminals must have at least one audio codec support. Table 8-3 describes the data stream rates of the G.xxx ITU-T Recommendations. [TB HEAD]Table 8-3 ITD-T audio codec recommendations [TB TXT] Audio Codec Recommendatio Bandwidth n Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-25 5/6/2010
  26. 26. G.711 64 kbps G.722 64, 56, 48 kbps G.723.1 5.3 and 6.3 kpbps G.728 16 kbps G.729a 8 kbps [END TBL] [C HD]Video Codec [BT C HD]A video codec encodes video from a camera for transmission on a transmitting H.323 terminal and decodes the received video code that is then sent to the video display on a receiving H.323 terminal. Because H.323 specifies the support of video as optional, the support of video codecs is optional as well. However, any H.323 terminal providing video communications must support video encoding and decoding as specified in the ITU-T H.261 recommendation. H.265 video compression standard was defined in 1990 by the CCITT (now the ITU-T) and was the first world-wide video compression standard. It was developed specifically to enable Video Conferencing units, manufactured by different companies, to communicate with each other, originally over ISDN or digital leased lines for operation at bit-rates between around 40 kbps and 1.9 Mbps. The application area is primarily interactive communications, and therefore the algorithm has been designed to provide good quality compression with a reasonably low delay. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-26 5/6/2010
  27. 27. [C HD]H.225 RAS (Registration, Admission, and Status) [BT C HD]RAS is the protocol between endpoints (terminals and gateways) and gatekeepers. RAS is used to perform registration, admission control, bandwidth changes, and status, and to uncouple procedures between endpoints and gatekeepers. An RAS channel is used to exchange RAS messages. This signaling channel is opened between an endpoint and a gatekeeper prior to the establishment of any other channels. [C HD]H.225 Call Signaling [BT C HD]H.225 is used to establish a connection between two H.323 endpoints. This is achieved by exchanging H.225 protocol messages on a call-signaling channel, which is opened between two H.323 endpoints or between an endpoint and the gatekeeper. [C HD]H.245 Control signaling [BT C HD]H.245 is used to exchange end to-end control messages governing the operation of the H.323 endpoint. These control messages carry information related to the following: capabilities exchange, the opening and closing of logical channels used to carry media streams, flow-control messages, and general commands. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-27 5/6/2010
  28. 28. [C HD]RTP (Real-Time Transport Protocol) [BT C HD]RTP is a transport protocol, it provides end-to-end delivery services of real-time audio and video. Whereas H.323 is used to transport data over IP-based networks, RTP is typically used to transport data via UDP. RTP together with UDP provides transport-protocol functionality. RTP provides payload-type identification, sequence numbering, timestamping, and delivery monitoring. UDP provides multiplexing and checksum services. RTP can also be used with other transport protocols. [C HD]RTCP (Real-Time Transport Control Protocol) [BT C HD]RTCP is the counterpart of RTP that provides control services. The primary function of RTCP is to provide feedback on the quality of the data distribution. Other RTCP functions include carrying a transport-level identifier for an RTP source, called a canonical name, which is used by receivers to synchronize audio and video. [B HD]SIP (Session Initiation Protocol) Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-28 5/6/2010
  29. 29. [BT B HD]SIP is a text-based signaling protocol used for creating and controlling multimedia sessions with two or more participants. It is a client-server protocol. SIP can interwork with gateways that pro- vide signaling protocols and media translations across dissimilar network segments such as PSTN to IP networks. SIP uses text-based messages, much like HTTP. SIP addressing is built around either a telephone number or a web host name. In the case of a web host name, the SIP address is based on a URL (uniform resource locator). The URL is translated into an IP address through a DNS (domain name server). SIP also negotiates the features and capabilities of the session at the time the session is established. Gateway Control Protocols [BT]The most immediate attraction to VoIP is to save money on long-distance transport. To date, it has been impractical to route VoIP "desktop to desktop," meaning interworking between PSTN and IP networks must be facilitated. This is done with a gateway. The two most applied gateways are the media gateway and the signaling gateway. Media gateways interconnect dissimilar networks. In this case, they connect the PSTN to IP networks. To do this successfully, they must mediate both signaling and transport between the two dissimilar networks (PSTN and IP). Media gateways coordinate call control and status. Gateway control protocols are signaling protocols. [B HD]MGCP Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-29 5/6/2010
  30. 30. [BT B HD]MGCP is the protocol used as an intermediary between MGC (Media Gateway Controller), also known as a call agent, and the media gateway. MGCP was developed by the IETF and details the commands and parameters that are passed between the MGC and the telephony gateway to be controlled. [BT]MGCP is a call control architecture where the call control intelligence is outside the gateway and is handled by external call control factors. The MGCP presupposes that these call control factors; known as call agents, will synchronize with each other to send commands to the gateways under their control. MGCP is a master/slave protocol, where the gateways are designed to execute commands sent by the call agents. [BT]The purpose of MGCP is to send commands from the call agent to a media gateway. MGCP defines both endpoints and connections. Endpoints are sources or sinks of data and can be either physical (terminating interfaces for a digital trunk or analog line) or virtual (a designated audio source). Endpoint identifiers have two components, the domain name of the gateway that is managing the endpoint, and a local name within that gateway. Examples of physical endpoints include interfaces on gateways that terminate: • a trunk connected to a PSTN switch (Class 5 or Class 4) • an analog POTS (Plain Old Telephone Service) connection to a telephone • a key system, Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-30 5/6/2010
  31. 31. • a PBX [BT]MGCP sends commands from the call agent to a media gateway. MGCP defines both endpoints and connections. [BT]Connections can be either point to point or multipoint in nature. A multipoint connection is estab- lished by connecting the endpoint to a multipoint session. For point-to-point connections, the endpoints of a connection could be in separate gateways or in the same gateway. [BT]The connections and calls are established by the actions of one or more call agents. The information communicated between call agents and endpoints can be either events or signals. An event would be a telephone going off hook, while a signal may be the application of a dial tone to an endpoint. These events and signals are grouped into packages, which are supported by a particular type of endpoint. One package may support events and signals for an analog line, while another package may support a group of events and signals for video lines. When media gateways interface with analog or PSTN connections to IP networks, MGCP will be the controlling protocol. [B HD]BICC (Bearer Independent Call Control) Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-31 5/6/2010
  32. 32. [BT B HD]BICC is a newer protocol used for media-gateway-to-media-gateway communications. It provides a means of supporting narrowband ISDN services across a broadband backbone network without impacting the interfaces to the existing N-ISDN network and end-to-end services. The BICC call control signaling protocol is based on N-ISUP signaling. [BT]In order for IP telephony networks to interoperate with the PSTN, they must interface with SS7. Softswitch solutions must include ISUP (ISDN User Part) and TCAP (Transaction Capabilities Application Part), ISUP is defined by ITU-T Q.761. Q.764 is the call control part of the SS7 protocol. ISUP is an SS7 protocol for signaling to set up and tear down circuit-switched voice calls between a softswitch/signaling gateway and an STP. ISUP determines the procedures for call setup and teardown on the SS7 network. [BT]TCAP is a peer protocol to ISUP in the SS7 protocol hierarchy for end-to-end signaling that not associated with call setup or specific trunks in the PSTN network. Two of its main uses are toll-free 800 number translation for routing and LNP (local number portability). TCAP provides services to many application parts. Common application parts include the INAP (Intelligent Network Application Part) and the MAP (Mobile Application Part). INAP is a signaling protocol used in the intelligent network architecture. It is part of the SS7 protocol suite, typically layered on top of the TCAP protocol. MAP is a mechanism for a Gateway to obtain a routing number for an incoming call. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-32 5/6/2010
  33. 33. [BT]We have now been introduced to the hardware and protocols that make up the VoIP architecture. We need to become familiar with how gateways (routers) and routing protocols interact to move packets through the multitude of networks that make up the Internet which is the transport for VoIP. [A HD]Routing Protocols [BT A HD]VoIP is routed over an IP network via routers. In order to deliver the best QoS, voice packets must be given priority over data packets. That means communicating to routers which packets have what priority. Router operations involve several processes. First, the router creates and continually updates routing tables gathering information from its neighbor routers to calculate the optimum path to the packet’s destination. Some table entries may be static established by a network administrator, but dynamic routing is considered a better technique as it constantly adapts to changing network conditions. The router determines the least cost (most efficient) path to each packet’s destination. [BT]Two algorithms are used to determine the least cost route. They are distance vector and link state. Protocols that make use of these algorithms are called IGPs (Interior Gateway Protocols). RIP (Routing Information Protocol) is an IGP utilizing a distance vector algorithm and the OSPF (Open Shortest Path First) protocol is an IGP utilizing a link state algorithm. At connection points where one network connects with another EGP (Exterior Gateway Protocol) is used. BGP (Border Gateway Protocol) is and EGP. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-33 5/6/2010
  34. 34. [B HD]RIP (Routing Information Protocol) [BT B HD]RIP is a distance vector protocol that uses hop count (the number of routers a packet passes through on the path to its destination) as a metric (a measure used to determine the least cost route). RIP is widely used as a routing protocol and is an IGP, performing routing within a single autonomous system. EGPs, such as BGP, perform routing between different autonomous systems. [B HD]OSPF (Open Shortest Path First) Protocol [BT B HD]The OSPF routing protocol was developed for IP networks by the IGP Working Group of the IETF. The Working Group was formed in 1988 to design an IGP based on the SPF (Shortest Path First) algorithm for use on the Internet. The need for OSPF was RIP’s inability to serve large internetworks. [BT]OSPF has two primary characteristics. The first is that the protocol is open, which means that its specification is in the public domain. The second principal characteristic is that OSPF is based on the SPF algorithm, referred to as the Dijkstra algorithm, named for the person credited with the algorithm. The OSPF specification is published as the IETF's RFC 1247. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-34 5/6/2010
  35. 35. [BT]OSPF is a link-state routing protocol that uses LSAs (Link State Advertisements) to communicate route changes to its near neighbors. Information on its attached interfaces, metrics, and other variables are included in OSPF LSAs. As OSPF routers receive link state information, they use the SPF algorithm to calculate their shortest path to other nodes within the network. [B HD]SPF Algorithm [BT B HD]The SPF routing algorithm is the basis for OSPF routing. When an SPF router is powered up, it initializes its routing-protocol data structures and then waits for indications from lower-layer protocols that its interfaces are functional. After a router is assured that its interfaces are functioning, it uses the OSPF Hello protocol to identify itself to its neighbors. All routers send hello packets to their neighbors and receive their neighbor’s hello packets. In addition to helping identifying neighbors, Hello packets also act as keep alive messages allowing their neighbor routers to know that they are still functional, alive. [BT]Each router sends LSAs to inform its neighbor routers when it detects a router’s state has changed. Failed routers can be detected quickly and the network's topology altered appropriately each time an LSA arrives from a neighbor. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-35 5/6/2010
  36. 36. [B HD]Border Gateway Protocol [BT B HD]BGP performs interdomain routing in TCP/IP networks. BGP is an EGP, which means that it performs routing between multiple autonomous systems or domains and exchanges routing information. BGP scales to Internet growth very efficiently. See Figure 8-6. BGP performs three types of routing: inter-autonomous system routing, intra-autonomous system routing, and pass-through autonomous system routing. [FIG]Figure 8-6 Interdomain routing Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-36 5/6/2010
  37. 37. AS 100 AS 200 BGP protocol must be running here connecting the three AS (Autonomous Systems) together AS 300 [C HD]Inter-autonomous system routing [BT C HD]Inter-autonomous system routing occurs between two or more BGP routers in different autonomous systems. Peer routers in these systems use BGP to maintain a consistent view of the internetwork topology. BGP neighbors communicating between autonomous systems must reside on the same physical network. The Internet serves as an example of an entity that uses this type of routing because it is comprised of autonomous systems or administrative domains. Many of these domains Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-37 5/6/2010
  38. 38. represent the various institutions, corporations, and entities that make up the Internet. BGP is frequently used to provide path determination to provide optimal routing within the Internet. [C HD]Intra-autonomous system routing [BT C HD]Intra-autonomous system routing occurs between two or more BGP routers located within the same autonomous system. Peer routers within the same autonomous system use BGP to maintain a consistent view of the system topology. BGP also is used to determine which router will serve as the connection point for specific external autonomous systems. Once again, the Internet provides an example of inter-autonomous system routing. An organization, such as a university, could make use of BGP to provide optimal routing within its own administrative domain or autonomous system. The BGP protocol can provide both inter- and intra-autonomous system routing services. [C HD]Pass-through autonomous system routing [BT C HD]Pass-through autonomous system routing occurs between two or more BGP peer routers that exchange traffic across an autonomous system that does not run BGP. In a passthrough autonomous system environment, the BGP traffic does not originate within the autonomous system in question and is not destined for a node in the autonomous system. BGP must interact with whatever Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-38 5/6/2010
  39. 39. intra-autonomous system routing protocol is being used to successfully transport BGP traffic through that autonomous system. [BT]BGP uses a single routing metric to determine the best path to a given network. This metric consists of an arbitrary unit number that specifies the degree of preference of a particular link. The BGP metric typically is assigned to each link by the network administrator. The value assigned to a link can be based on any number of criteria, including the number of autonomous systems through which the path passes, delay, speed, stability, or cost. [B HD]RSVP (Resource Reservation Protocol) [BT C HD]RSVP is a network control protocol that enables Internet applications to obtain special QoS (qualities of service) for their data streams. [BT]RSVP is not a routing protocol it works in conjunction with routing protocols and installs the equivalent of dynamic access lists along the routes that routing protocols calculate. RSVP is a transport protocol in the OSI Model seven-layer protocol stack. [BT]In RSVP, a data flow is a sequence of messages that have the same source, destination and QoS. QoS requirements are communicated through a network via a flow specification, which is a data Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-39 5/6/2010
  40. 40. structure used by internetwork hosts to request special services from the internetwork. A flow specification can guarantee how the internetwork will handle portions of its traffic. RSVP supports three traffic types: • best-effort • rate-sensitive • delay sensitive. [BT]The type of data-flow service used to support these traffic types depends on the QoS implemented. The following paragraphs describe these traffic types and associated services. Best-effort Traffic Best effort is traditional IP traffic. The service supporting best-effort traffic is called best-effort service. Rate-sensitive traffic Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-40 5/6/2010
  41. 41. [BT]Rate-sensitive traffic sacrifices timeliness for guaranteed rate. Rate-sensitive traffic might request 100 Kbps of bandwidth. If it actually sends 200 Kbps for an extended period, a router can delay traffic. Rate-sensitive traffic is not intended to run over a circuit switched network; however, it usually is associated with applications that are ported from a circuit-switched network (such as ISDN) to a datagram network (IP). [BT]H.323 videoconferencing is an example of such an application. It is designed to run on ISDN (H.320) or ATM (H.310). H.323 encoding is a nearly constant rate, and it requires a constant transport rate. The RSVP service supporting rate sensitive traffic is called guaranteed bit-rate service. Delay- sensitive traffic is traffic that requires timeliness in delivery and varies its rate accordingly. MPEG-II video, for example, averages about 3 to 7 Mbps depending on the amount of change in the picture. For example, 3 Mbps might be required for a picture of a painted wall, but 7 Mbps might be required for a picture of waves on the ocean. MPEG-II video sources send key and delta frames. Typically, 1 or 2 key frames per second describe the whole picture, and 13 or 28 delta frames describe the change from the key frame. Delta frames are usually substantially smaller than key frames. As a result, rates vary quite a bit from frame to frame. A single frame, however, requires delivery within a frame time (the time allotted to a particular frame type) or the codec will be unsuccessful in delivering timely packets. A specific priority must be negotiated for delta-frame traffic. RSVP services supporting delay-sensitive traffic are referred to as controlled-delay service (non real-time service) and predictive service (real- time service). Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-41 5/6/2010
  42. 42. [BT]In the context of RSVP, QoS is an attribute specified in flow specifications that is used to determine the way in which data interchanges are handled by routers, receivers, and senders. RSVP is used to specify the QoS by both hosts and routers. Hosts use RSVP to request a QoS level from the network on behalf of an application data stream. Routers use RSVP to deliver QoS requests to other routers along the path of the data stream. This results in RSVP maintaining the router and host state to provide the requested service. [B HD]Transport Protocols [BT B HD]Figure 8-7 illustrated the actual VoIP conversation taking place between the two IP phones and no longer involves the Media Gateway. Once the called party’s phone is given the ring signal the media gateway is no longer needed. The media gateway was involved only during call setup. The following paragraphs describe RTP the Realtime Transport Protocol that carries the voice conversation. [FIG]Figure 8-7 Signaling separate from VoIP converstion Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-42 5/6/2010
  43. 43. Media Gateway 1. Off hook 2. Send Dial tone 4. is phone on hook? Yes 3. Dial number 5 .Give caller ip address 6 Receive IP 7. set up UDP port Call setup 8. Ring signal signaling Signaling uses TCP/IP Conversation uses UDP UDP UDP Called party Calling party UDP Real time Transport Protocol VoIP Conversation [C HD]RTP (Real-Time Transport Protocol) [BT C HD]RTP is the most popular of the VoIP transport protocols. It is specified in RFC 1889 under the title of "RTP: A Transport Protocol for Real Time Applications." This RFC actually describes both RTP and another protocol, RTCP. These two protocols are necessary to support real-time applications like voice and video. RTP operates on the layer above UDP, which does not avoid packet loss or guarantee the correct order for the delivery of packets. RTP including sequence numbers that help applications using RTP to detect lost packets and ensure packet delivery in the correct order. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-43 5/6/2010
  44. 44. [BT]RTP packets include a timestamp that gives the time when the packet is sampled from its source media stream. This timestamp assists the destination application with determining a synchronized playout to the destination user and to calculate delay and jitter, two very important detractors of voice quality. RTP does not have the capacity to correct delay and jitter, but it does provide additional information to a higher-layer application so that the application can make determinations as to how a packet of voice or data can best be handled. [BT]RTCP provides a number of messages that are exchanged between session users provide feedback regarding the quality of the session. The type of information includes details such as the numbers of lost RTP packets, delays, and inter-arrival jitter. As voice packets are transported in RTP packets, RTCP packets transfer quality feedback. Whenever an RTP session opens, an RTCP session is also opened. That is, when a UDP port number is assigned to an RTP session for the transfer of media packets, another port number is assigned for RTCP messages. The digitally encoded voice by taking one or more digitally encoded voice samples and attaching an RTP header forms an RTP packet, which consists of an RTP header and a payload of the voice samples. These RTP packets receive a UDP header. This combination then goes to IP where an IP header is attached and the resulting IP datagram is routed to the destination. At the destination, the headers are used to pass the packet up the stack to the appropriate application. [D HD]RTP Headers Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-44 5/6/2010
  45. 45. [BT D HD]The RTP payload is comprised of digitally coded voice samples. An RTP header is attached to this payload and the packet is passed to the UDP layer. The RTP header contains the necessary information for the destination to reconstruct the original voice sample. [D HD]RTP Control Protocol (RTCP) [BT D HD]RTCP enables exchanges of control information between session participants with the goal of providing quality-related feedback. This feedback is used to detect and correct distribution issues. The combination of RTCP and IP multicast enables a network operator to monitor session quality. RTCP provides information on the quality of an RTP session. RTCP empowers network operators to obtain information about delay, jitter, and packet loss. and to take corrective action to improve quality. RTP addresses Payloads and RTP carries the Payload. [B HD]IPv6 [BT B HD]The previous discussion assumed the use of IPv4 (Internet Protocol version 4), the predominant version of IP in use today. A new version, IPv6, is now coming on the market. The explosion of Internet addresses necessitates the deployment of IPv6. IPv6 makes possible infinitely more addresses than IPv4. Enhancements offered by IPv6 over IPv4 include: Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-45 5/6/2010
  46. 46. • Expanded address space. Each address is allocated 128 bits instead of 32 bits in IPv4. • Simplified header format. This enables easier processing of IP datagrams. • Improved support for headers and extensions. This enables greater flexibility for the introduction of new options. [BT]Flow-labeling capability enables the identification of traffic flows for real-time applications. Authentication and privacy Support for authentication, data integrity, and data confidentiality are supported at the IP level rather than through separate protocols or mechanisms above IP. [BT]This chapter, so far, has addressed the building blocks of VoIP. It will be necessary in future chapters to understand the concepts and protocols contained in this chapter. For example the following analogy is quite important when comparing PSTN and IP voice: Just as the PSTN and softswitch networks can be broken down into the three elements of access, switching, and transport, VoIP can be summarized as a study of three types of protocols: signaling, routing, and transport. Although protocols will continue to evolve and new protocols will emerge, those addressed in this chapter will constitute the predominant structure of softswitch architecture for the next few years. [BT]H.323 is not the only protocol that can bring us VoIP. A signaling protocol, SIP, is making inroads and challenging H.323 for market share. We will throughout the balance of this chapter explore SIP and at the end contrast its differences with H.323 [A HD]SIP a Signaling Protocol Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-46 5/6/2010
  47. 47. [BT A HD]If the PSTN (Public Switched Telephone Network) were to be replaced, “cold turkey” (take the PSTN out and replace it overnight), the best option available today would be based on VoIP (Voice over IP) and the SIP (Session Initiation Protocol). Keep it in mind that “cold turkey” is not an acceptable option even in a small environment. Much of the VoIP industry has targeted solutions that leverage existing circuit-switched infrastructure, like VoIP gateways that interface a PBX (private branch exchange) and an IP (Internet Protocol) network. SIP is an architecture that offers more features than a circuit-switched network will ever be able to offer. [BT]SIP is a signaling protocol. It uses a text-based syntax similar to HTTP (Hypertext Transfer Protocol) used in web access today. Programs that are designed for the parsing (resolving into component parts) HTTP can be adapted easily for use with SIP. SIP addresses, known as SIP URLs (uniform resource locators) are in the form of web addresses. A web address is the equivalent of a telephone number in an SIP network. In addition, PSTN phone numbers can be incorporated into an SIP address for interfacing with the PSTN. An email address is portable so you can check your email from an Internet-connected terminal from anyplace in the world. Regular POTs telephone numbers are not portable and they only ring at one physical location. SIP offers a mobility function that can follow subscribers to whatever phone they are nearest to at a given time called, “follow-me”. [BT]Like H.323, SIP handles the setup, modification, and teardown of multimedia sessions, including voice. SIP was designed as a part of the IETF (Internet Engineering Task Force) multimedia data and Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-47 5/6/2010
  48. 48. control architecture. It is designed to interwork with other IETF protocols such as the SDP (Session Description Protocol), RTP, and the SAP (Session Announcement Protocol). It is described in the IETF's RFC 2543. Many consulting firms in the VoIP and softswitch industry believe that SIP will replace H.323 as the standard signaling protocol for VoIP. [BT]SIP is part of the IETF standards process and is modeled on other Internet protocols such as the SMTP (Simple Mail Transfer Protocol) and HTTP. It is used to establish, change, and tear down calls between one or more subscribers in an IP-based network. In order to provide telephony serviceably a number of different standards and protocols must come together specifically to: [BEG BL] • Ensure transport - RTP • provide signaling with the PSTN • guarantee voice quality - RSVP (Resource Reservation Setup Protocol) • provide directories - LDAP (Lightweight Directory Access Protocol) • authenticate users - RADIUS (Remote Access Dial-In User Service) • scale to meet anticipated growth expectations. [END BL] [B HD]What Is SIP? Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-48 5/6/2010
  49. 49. [BT B HD]SIP is architected on two classes of network entities: clients, called UAs (user agents), and servers, basically a client/server architecture. VoIP uses SIP to originate at a client and terminates at a server. Types of clients in the technology currently available for SIP telephony include a PC loaded with a telephony agent (softphone) or an SIP telephone. Clients can in fact reside on the same platform as a server. For example, a PC on a corporate WAN (wide area network) might be the server for the SIP telephony application, but it may also be used as a user's telephone (a client). We will take a look at UAs and their counterparts UA Clients and UA Servers in the following sections. [C HD]SIP Architecture [BT C HD]SIP is a client/server architecture. The client in this architecture is the UA, which interacts with the user. It usually has an interface in the form of a PC or an IP phone (SIP phone). Four types of SIP servers exist. The type of SIP server used determines the architecture of the network. Those servers are the UAS (user agent server), the redirect server, the proxy server, and a registrar. [BT]A SIP device can function as both a UAC (UA client) and as a UA server. As a UAC, the SIP device can initiate SIP requests. As a UA server, the device can receive and respond to SIP requests. As a standalone device, the UA can initiate and receive calls enabling SIP to be used for peer-to-peer communications. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-49 5/6/2010
  50. 50. [BT]The function of SIP is best understood via the HTTP model upon which it is based. SIP is a request/response protocol. A client is a SIP device that generates a request. A server is an SIP device that receives requests and returns responses. When a web site is requested, the client provides a request by typing in the URL, such as http://www.course.com. The server upon which the web site is hosted responds with, Thomson Course Technology's web page. SIP uses a similar procedure. The UA initiates a request known as a UAC, and the UA that returns the response is the UAS. This overall action is known as a SIP transaction. The steps in initiating a SIP call are described next. INVITE The first message sent by a calling party is to invite users to participate in a session. The message contains information in the SIP header that identifies: • the calling party • caller ID • called party • call and sequence numbers. [BT]It indicates a call is being initiated. When a multiple choice of SDP parameters is offered, the ones chosen are returned along with the success code (200) in the response message. Code (200) is a response of “OK”. See Table 8-4 for an expanded list of codes. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-50 5/6/2010
  51. 51. [TB HEAD]Table 8-4 IP number codes and descriptions [TB TXT] Number Number Description Description code code 100 Trying 413 Request, entity too large 180 Ringing 414 Request, DRI too large 181 Call is being forwarded 415 Unsupported media type 182 Queued 420 Bad extension 183 Session progress 480 Temporarily not available 200 OK 481 Call leg/transaction does not exist 202 Accepted 482 Loop detected 300 Multiple choices 483 Too many hops 301 Moved permanently 484 Address incomplete 302 Moved temporarily 485 Ambiguous 305 Use proxy 486 Busy here 380 Alternative service 487 Request cancelled 400 Bad request 488 Not acceptable here 401 Unauthorized 500 Internal severe error 402 Payment required 501 Not implemented 403 Forbidden 502 Bad gateway 404 Not found 503 Service unavailable 405 Method not allowed 504 Gateway timeout 406 Not acceptable 505 SIP version not supported 407 Proxy authentication 600 Busy everywhere 408 Request timeout 603 Decline 409 Conflict 604 Does not exist anywhere 410 Gone 606 Not acceptable 411 Length required [END TBL] Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-51 5/6/2010
  52. 52. ACK [BT]An ACK is used to acknowledge the reception of a final response to an INVITE. A client originating an INVITE request issues an ACK request when it receives a final response for the INVITE, completing a three-way handshake. OPTIONS [BT]This queries a server about the called endpoint’s capabilities, including which methods and which SDPs it supports. This determines the media types a remote user supports before placing the call. BYE request from either party that ends the call, initiated by placing the telephone on hook. CANCEL Cancel is used to abandon a session. In two-party sessions, termination by one of the parties implies that the session is terminated. This method cancels pending transactions and identifies the call via the call ID, call sequence, and To and From values in the SIP header. This data is useful in billing and statistics. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-52 5/6/2010
  53. 53. REGISTER Register requests are sent by IP devices to inform a server about their current location. SIP servers are co-located with SIP registrars. This enables the SIP server to find an IP device. [BT]When the call arrives at the remote endpoint, the phone rings and a new response is sent to that endpoint: RINGING (180). This is analogous to the Q.931 ALERTING message used in the PSTN. The time between the caller dialing the last digit and the time RINGING is received by the caller is known as the PDD (Post Dial Delay) for SIP call setup. If a telephone number is involved in addressing the call, the numbers must be translated into an IP address. Table 8-5 provides a comparison of SIP and PSTN signals. [TB HEAD]Table 8-5 SIP and PSTN signal comparison [TB TXT] PSTN SIP Sip Code Q.931 Call Proceeding TRYING 100 Q.931 Alerting RINGING 180 Q.931 Connect ACK 201 Q.931 Connect XINVITE Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-53 5/6/2010
  54. 54. [END TBL] [BT]When the called party answers the phone, a response is sent to the calling party's UA. The UA sends another request, ACK, acknowledging the response to the INVITE request. At that moment, media begins to flow on the transport addresses of the two endpoints. The ACK may carry the final SDP parameters for the media type and format provided by the receiving endpoint. The sequence of the INVITE and following ACK messages is similar to the Q.931 CONNECT message. ACKs do not require a response. Table 8-5 displays the response codes for SIP. [BT]At this point in the call sequence with the RTCP (Real- Time Control Protocol) providing the monitoring of the quality of the connection and its associated statistics. Next a BYE request from either party ends the call. As all messages are sent via UDP, no further action is required. A BYE request is initiated when a telephone goes on-hook. [B HD]SIP Calls via a Proxy Server Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-54 5/6/2010
  55. 55. [BT B HD]Proxy servers with respect to SIP are similar in function to proxy servers that serve a web site (mail relay via SMTP) for a corporate LAN. A SIP client in this case would send a request to the proxy server that would either handle it or pass it on to another proxy server that would take the call. The secondary servers would see the call as coming from the client. By virtue of receiving and sending requests, the proxy server is both a server and a client. Proxy servers function well in call forwarding and follow-me services. [BT]Proxy servers can be classified by the amount of state (condition at the time) information they store in a session. SIP defines three types of proxy servers: call stateful, stateful, and stateless. Call stateful proxies need to be informed of all the SIP transactions that occur during the session and are always in the path taken by SIP messages traveling between end users. These proxies store state information from the moment the session is established until the moment it ends. A stateful proxy is sometimes called a transaction stateful proxy as the transaction is its sole concern. A stateful proxy stores a state related to a given transaction until the transaction concludes. It does not need to be in the path taken by the SIP messages for subsequent transactions. Forking proxies are good examples of stateful proxies. Forking proxies are used when a proxy server tries more than one location for the IP device, it "forks" the invitation. [BT]Now that we acquired base knowledge regarding H.323 and SIP let us contrast the two. The following sections will explore the complexities both SIP and H.323 and provide a general comparison between them. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-55 5/6/2010
  56. 56. [A HD]Comparison of SIP to H.323 [BT A HD]H.323 is not a protocol by itself, but rather defines how to use other protocols in a specific manner to create a service. H.323 was developed by the ITU (International Telecommunications Union) in the mid-1990s and consists of several protocols, including H.225 RAS (registration, admission, and status) signaling, H.225.0 call signaling, H.245 control signaling, and RTP. It also includes several standards for voice and video digitization and compression. [BT]H.323 was originally designed to implement multimedia conferences on a LAN. In its original form, it was intended that certain conferences would be announced, or advertised in advance, and interested parties would subscribe or register to participate in the conference. These conferences may be interactive or may be broadcast events with participants viewing or listening only. H.323 and SIP have four fields of comparison: complexity, extensibility, scalability, and services. You are about to about to discover these characteristics. [B HD]Complexities of H.323 Versus SIP [BT B HD]As VoIP protocols evolve, they become more efficient. If simplicity encourages acceptance, SIP is a marked improvement over H.323 primarily due to its greater simplicity, which Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-56 5/6/2010
  57. 57. translates to greater reliability. Given its simplicity relative to H.323, SIP enjoys a faster call setup, a prerequisite for carrier-grade voice applications. H.323 is a rather complex protocol. [BT]A basic but interoperable SIP Internet telephony implementation can get by with four headers (To, From, Call-ID, and CSeq) and three request types (INVITE, ACK, and BYE) and is small enough to be completed in a days time or less. A fully functional SIP client agent with a GUI (Graphical User Interface) can save many man hours, weeks, or even months of work. [BT]H.323 uses a binary representation for its messages. SIP, on the other hand, encodes its messages as text, similar to HTTP and the RTSP (Real-Time Streaming Protocol). This leads to simple parsing and generation, particularly when done with powerful text-processing languages. The textual encoding also simplifies debugging, allowing for manual entry and perusing messages. Its similarity to HTTP also allows for code reuse; existing HTTP parsers can be easily modified for SIP usage. [BT]A huge criticism of H.323 is that it can result in the transmission of many unnecessary messages across the network. This causes H.323 to not scale well. The overhead on large networks associated with handling a large number of messages has a significant effect on system performance. A more heavily loaded system will result in poor voice quality. QoS measures do not adequately address network traffic and call setup issues in H.323. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-57 5/6/2010
  58. 58. [BT]H.323's complexity also stems from its use of several protocol components. There is no clean separation of these components; many services require interactions between several of them. (Call forward, for example, requires components of H.450, H.225.0, and H.245.) The use of several different protocols also complicates firewall traversal. Firewalls must act as application-level proxies, to parsing the entire message to arrive at the required fields. The operation is stateful since several messages are involved in call setup. SIP, on the other hand, uses a single request that contains all necessary information. [BT]An additional aspect of H.323's complexity is its duplication of some of the functionality present in other parts of the protocol. In particular, H.323 makes use of RTP and RTCP. RTCP has been engineered to provide various feedback and conference control functions in a manner that scales from two-party conferences to thousand-party broadcast sessions. H.245, however, provides its own mechanisms for both feedback and simple conference control (such as obtaining the list of conference participants). These H.245 mechanisms are redundant and have been engineered for small to medium sized conferences only. [B HD]Scalability Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-58 5/6/2010
  59. 59. [BT B HD]H.323 and SIP differ in terms of scalability. The three main concerns are managing a large number of domains, server-processing capabilities, conferencing, and feedback. H.323 was originally conceived for use on a single LAN and not a large number of domains. Issues such as wide area addressing and user location were not a concern. The latest version defines the concept of a zone and defines procedures for user locations across zones for email names. However, for large numbers of domains, and complex location operations, H.323 has scalability problems. It provides no easy way to perform loop detection in complex multidomain searches (it can be done statefully by storing messages, which is not scalable). SIP, however, uses a loop detection algorithm similar to one used in BGP, which can be performed in a stateless manner. [BT]Another factor for scalability is server processing. Regarding server processing in an H.323 system, both telephony gateways and gatekeepers will be required to handle calls from a multitude of users. Similarly, SIP servers and gateways will need to handle many calls. For large, backbone IP tele- phony providers, the number of calls being handled by a large server can be significant. In SIP, a transaction through several servers and gateways can be either stateful or stateless. In the stateless model, a server receives a call request, performs some operation, forwards the request, and completely forgets about it. SIP messages contain sufficient state to enable the response to be forwarded correctly. [BT]Because the protocol is simpler, SIP requires less code to implement than H.323. This is reflected in lower fixed (code space) and dynamic memory requirements for both the protocol stack itself and the host application. Because of the smaller number of messages that the processor needs to handle to Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-59 5/6/2010
  60. 60. set up a call, SIP is faster than H.323. This means that SIP can process more calls per second than H.323 and that SIP requires a less powerful CPU to handle the same number of calls. The result is in a higher number of calls handled for a given system's usage. This results in larger message sizes for SIP than for H.323. However, it is better to send a few large frames of data than a lot of small frames, except for slow speed links. [BT]SIP can be carried on either TCP (Transmission Control Protocol) or UDP. In the case of UDP, no connection state is required. This means that large, backbone servers can be based on UDP and operate in a stateless fashion, reducing significantly the memory requirements and improving scalability. H.323, on the other hand, requires gatekeepers (when they are in the call loop ) to be stateful. They must keep the call state for the entire duration of a call. Furthermore, the connections are TCP based, which means a gatekeeper must hold its TCP connections for the entire duration of a call. This can pose serious scalability problems for large gatekeepers. [BT]Conferencing is another concern for scalability. H.323 supports multiparty conferences with multicast data distribution. However, it requires a central control point (called an MC) for processing all signaling, even for the smallest conferences. This presents several difficulties. First, should the user providing the MC functionality leave the conference and exit their application, the entire conference terminates. In addition, since MC and gatekeeper functionality is optional, H.323 cannot support even three party conferences in some cases. MC is a bottleneck for larger conferences. Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-60 5/6/2010
  61. 61. [BT]H.323 Version 2 defined the concept of cascading MCs, allowing for a very limited, application- layer, multicast distribution tree of control messaging. This improves scaling somewhat, but for even larger conferences; the H.332 protocol defines additional procedures. This means that three distinct mechanisms exist to support conferences of different sizes. SIP, however, scales to all different conference sizes. There is no requirement for a central MC; conference coordination is fully distributed. This improves scalability and complexity. Since it can use UDP as well as TCP, SIP supports native multicast signaling, enabling a single protocol to scale from sessions with two to millions of members. Table 8-6 compares the SIP and H.323 call features. A number of these features will not be discussed, those that may be familiar to you would be; hold (place a call on hold), Forward (forward a call), and blind transfer (transferring a call). The complete list would be hundreds of features long. Feedback is another concern when comparing H.323 and SIP. H.245 defines procedures that enable receivers to control media encodings, transmission rates, and error recovery. This kind of feedback makes sense in point-to-point scenarios, but it ceases to be functional in multipoint conferencing. SIP, instead, relies on RTCP for providing feedback on reception quality (and also for obtaining group membership lists). RTCP, like SIP, operates in a fully distributed fashion. The feedback it provides automatically scales from a two-person point-to-point conference to huge broadcast style conferences with millions of participants. [TB HEAD]Table 8-6 Comparing SIP and H.323 features Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-61 5/6/2010
  62. 62. [TB TXT] Feature SIP H.323 Blind transfer Yes Yes Operator-assisted transfer Yes No Hold Yes, through SDP No Multicast conferences Yes Yes Multiunicast conferences Yes Yes Bridged conferences Yes Yes Forward Yes Yes Call park Yes No Directed call pickup Yes No [END TBL] [B HD]Extensibility [BT B HD]Extensibility is a key metric for measuring an IP telephony signaling protocol. Telephony is a tremendously critical service, and Internet telephony is poised to displace the existing circuit- switched infrastructure developed to support it. As with any heavily used service, the features it provides have evolved over time as new applications developed. This makes compatibility among versions a complex issue. As the Internet is an open, distributed, and evolving service; you can expect extensions to IP telephony protocols will be extensive. [BT]To enhance extensibility; numerical error codes are hierarchically organized, as in HTTP. Six basic classes exist, each of which is identified by the hundreds digit in the response code (refer to Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-62 5/6/2010
  63. 63. Table 8-5). Basic protocol operation is dictated solely by the class, and terminals need only understand the class of the response. The other digits provide additional information, usually useful but not critical. This allows for additional features to be added by establishing semantics for the error codes in a class while achieving compatibility. The header fields are self describing due to the textual encoding. [BT]H.323 requires full backwards compatibility for each progressive version. A critical issue for extensibility are: audio and video codecs (coder/decoders). Hundreds of codecs have been developed, many of which are proprietary. SIP uses the SDP to convey the codecs supported by an endpoint in a session. This is due to the dissimilarities of ASN versus text. [BT]SIP can work with any codec. With H.323, each codec must be centrally registered and standardized. SIP enables new services to be defined using only a few powerful third-party call control mechanisms. These mechanisms can be used to construct a variety of services, including blind transfers, operator assisted transfers, three-party calling, bridged calling, dial-in bridging, multiunicast to multicast transitions, ad hoc bridge invitations and transitions, and various forwarding variations. It is not important for you to understand these mechanisms at this time, as they will become important and explained in detail when you take more advanced courses in VoIP. [BT]A very important aspect of extensibility is modularity. Internet telephony requires a large number of various functions; these include basic signaling, conference control, QoS, directory access, service Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-63 5/6/2010
  64. 64. discovery, and so on. Placing these functions in modules allows modification, insertion, or extraction without impacting other functions. [BT]H.323 is less modular than SIP. It defines a vertically integrated protocol suite for a single application. The blend of services provided by the H.323 includes: capability exchanges, conference control, maintenance operations, basic signaling, QoS, registration, and service discovery. These services are interwoven within the various subprotocols of H.323. SIP's modularity enables it to be used in conjunction with H.323. A user can use SIP to locate another user, taking advantage of its rich multihop search facilities. When the user is finally located, he or she can use a redirect response to an H.323 URL, indicating that the actual communication should take place with H.323.18 Table 8-7 compares H.323's call control service with that of SIP. [TB HEAD]Table 8-7 Comparing H.323 control service with SIP [TB TXT] Criteria H.323 SIP Complexity Very complex Simple Message set Many messages Few messages Debugging Have to alter tools when a Simple tools protocol is extended Extensibility Extensible Very extensible User extendable ASN.l-complex Text based-easy Elements that must Clients, gatekeepers, MCU, maintain states gateways, UAs, some proxy servers Convergence+ Guide To Convergence Technologies ISBN# 0619131179 Page 1-64 5/6/2010

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