.. the workload. ATM is one technology that definitely does just that. Beginning at the way the header is made up; ATM is arguably the best choice for transporting voice. The header, itself contains a pointer, which allows a digital signal level 0 (DS0) structure to be maintained.
DS0 are the lines that today transport voice. They are multiplexed together to get larger and larger number of signals through. Signaling with VoATM is compared in the pictures below. VoATM has the ability to either transport voice signals transparently through the network or to interpret and move the signals at ATM speeds. The second is more advantageous because of the use of SVCs or Switched Virtual Circuits.
These are circuits, which do not have a physical end-to-end connection between users established. Rather, signals are passed through the network along a logical path that works exactly the same as if a sold connection was there. Allowing VoATM signaling translation is better for three reasons: SVCs are more efficient users of bandwidth than PVCs. QoS for connections do not need to be constant, as with PVCs. The ability to switch calls within the network can lead to the elimination of the tandem private branch exchange (PBX).
The addressing used for VoATM is 20 bytes in length and supports both public and private addressing schemes. Routing is handled by Private Network to Network Interface (PNNI) protocol. Newtons Telecom dictionary describes PNNI as an extremely scalable, full function, dynamic, multi-vendor protocol. The way it works is a virtual circuit (VC) request a connection with a certain QoS through the ATM network. The source ATM switch goes out and finds the best route matching the QoS requested.
Each switch along the path is checked to determine if it has the appropriate resources necessary. When the connection is established, voice traffic flows between end stations as if a leased line existed between the two. VoATM has many built in feature for controlling delay and delay variance. The VCs can request specific bit rates with bandwidth and delay guarantees. There are also VC queues allowing each traffic stream to be treated uniquely.
The use of small, fixed-size cells reduces queuing delay and delay variation due to variable-size packets. Voice over Frame Relay Frame Relay is one of the most widely implemented WAN technologies. Its inexpensive, yet reliable track record has made it very popular. The signaling used for Frame Relay has been historically proprietary. This has inhibited its progress into the voice market.
The Frame Relay forum however has developed a set of standards known collectively as FRF.11 for VoFR (Voice over Frame Relay). Static tables handle addressing for VoFRcertain dialed digits choose which PVC to use. Voice is routed depending on the protocol chosen for establishing PVCs. Depending on the protocol such things as bandwidth limits, hops, delay or some combination can determine route, although most concentrate mainly on bandwidth utilization. When it comes to preventing delay Frame Relay falls a bit short. The frame size is variable. This means that delay variance is also variable. Different size frames pass through networking devices at different speeds. The smaller the frame the faster the passing but its an inefficient use of bandwidth because of the extra information associated with each frame.
Longer frames take considerably longer but because more information is encapsulated within each frame its a better use of bandwidth. Up until now, the solution to this problem has been proprietary. However, the Frame Relay Forum is defining what is known as FRF.12, which will create an industry standard to solve the small frame size problem. Voice over IP Whats so different about Voice over IP rather than VoATM or VoFR? VoIP is capable of converging voice and data at the application layer, rather than manipulating lower layers. This has the most appeal to people interested in cable, DSL and wireless networks because it allows service providers to bundle their services.
To make this a bit clearer, the protocols running over the network itself control all the functionality instead of the network itself. Regardless of the technology running under it VoIP provides a solution for everyone. In order to do this VoIP must provide a solution for signaling, routing and addressing. Signaling for VoIP has three distinct areas: PBX to router, router to router and router to PBX. The corporate intranet, to the PBX, looks like a trunk line.
Signals are sent from the PBX through the corporate intranet to seize a trunk using any of the common signaling methods. FXS or E&M signaling is used for fax and in the future common channel signaling (CCS) or Q.SIG will become available as a digital signaling method. The PBX then forwards the dialed digits to the router in the same way they would be sent to a Telco switch. Within the router, the digits are mapped to an IP address and using Q.931 call setup establishes a request to the remote address. Meanwhile, the control channel is used to set up the Real-Time Control Protocol (RTCP) audio streams and Resource Reservation Protocol (RSVP) is used to guarantee QoS. When the remote router receives the Q.931 call request, it signals a line seizure to the PBX. After the PBX acknowledges, the router forwards the dialed digits to the PBX and signals a call acknowledgment to the originating router.
All the responsibility for session establishment and signaling is with the end stations. To successfully accomplish this, additional enhancements must be made to the signaling stack. H.323 is such an addition and will be discussed in-depth next. Corporations should already have an IP addressing scheme in place. The voice interfaces will show up as additional nodes, either as an extension of the existing scheme or with new IP addresses.
The dial plan mapper performs translation of these addresses. The destination telephone number or some portion is mapped to the destination IP address. When the number is received from the PBX, the router compares the number to those mapped in the routing table. If a match is found, the call is routed to the IP host and is transparent to the user. VoIP real strength is rooted in IPs mature and sophisticated routing protocols.
By using routing protocols such as Enhanced Interior Gateway Routing Protocol (EIGRP) specific factors including delay are taken into consideration for best route decisions. Other advanced features like policy routing and access-list allow you to create highly secure networks. Increasing innovations, such as tag switching, are also being developed to allow better traffic engineering. This will lead to the ability to shift traffic load based on different variables, such as time of day. Traditionally, IP traffic has been handled on a best effort mechanism. Traffic was first come, first serve but voice is not tolerant to retransmission and delay.
Also the variable packet size problem is an issue. Once again using RSVP to initially find a route through network and then using RFC 1717 to break up the large packet to a standard, smaller size was the solution. Weighed fair queuing was also used to put different traffic types into specific QoS queues and thus reducing queuing delay. H.323 The ITU created the H.323 standard to enable mixed-media communications over packet based networks that do not provide QoS. The standard is said to be an umbrella encompassing various associated standards (See chart 2).
Although H.323 provides support for audio, video, data and multipoint conferencing, only the audio support is mandatory. H.320 H.321 H.322 H.323 H.324 Purpose Narrowband ISDN Broadband ISDN, LAN, ATM Guaranteed bandwidth packet networks No guaranteed bandwidth packet networks and Ethernet Analog PSTN telephone system Audio G.711, 722, 728 G.711, 722, 728 G.711, 722, 728 G.711, 722, 723, 728, 729 G.723 Video H.261, 263 H.261, 263 H.261, 263 H.261, 263 H.261, 263 Multipoint H.231, 243 H.231, 243 H.242, 243 H.323 Control H.320, 242 H.242 H.231, 243 H.245 H.245 Interface I.400 AAL I.400, TCP/IP UDP/IP, TCP/IP V.34 -The Irwin Handbook of Telecommunications, 4th edition. H.323 power comes from its multitude of other standards. Many applications are possible by using this architecture including: Internet telephony, desktop videoconferencing, LAN telephony, conference calling and mixed media conferences such as voice, video and whiteboard. Interoperability is a key feature in todays networks. H.323 uses industry open standards which when followed by vendors allows other products to work together.
A general H.323 architecture is shown in figures 1 & 2 below. The TCP/IP network uses TCP (reliable connection-oriented protocol) for call setup and UDP (fast, connection-less protocol) for voice packets. A signaling channel known as the RAS channel is used for communications between devices. Real-Time Transport (RTP) is used to sequence packets, compensating for UDPs lack of this capability. Real-Time Control Protocol (RTCP) monitors QoS. -Figure 1-Radcom VoIP Technology Protocol Reference poster. Gatekeeper Manages a zone (collection of H.323 devices).
o Required Functionality Address translation, admissions and bandwidth control. o Optional Functionality Call authorization, bandwidth management, supplementary services, directory services, call management services. Gateway Provides interoperability between different networks, converts signaling and media e.g. IP/PSTN gateway H.323 Terminal Endpoint on a LAN. Supports real-time, 2-way communications with another H.323 entity.
Must support voice (audio codecs) and signaling (Q.931, H.245, RAS). Optionally supports video and data e.g. PC phone or videophone, Ethernet phone. MCU Supports conferences between 3 or more endpoints. Contains multipoint controller (MC) for signaling. May contain multi-point processors (MP) for media stream processing. Can be stand-alone (i.e.
PC) or integrated into a gateway, gatekeeper or terminal. Implementations Types of VoIP Implementations VoIP through a Router Benefits: If a PBX already exists, it makes maximum use of existing resources The service is completely transparent to users The connection can be completed over any available packet network. Blockage of voice calls should be rare since the PBX can complete the call over the PSTN. LAN TELEPHONES This configuration allows you to connect devices directly to the network. Analog telephones can be connected using an Ethernet adapter through a PC. The PC gives you a lot of versatility because it can substitute for the telephones button interface.
Calls within the zone are controlled by the VoIP gateway rather than having a PBX onsite. This implementation is inexpensive and great for branch offices. IP PBX Also known as the un-PBX. This implementation has PBX hardware and software function loaded on a PC running something like Windows NT or Unix. The various cards can be loaded into the PC and generate call-processing programs.
Obviously, though, the fault-tolerance of an un-PBX compared to a real PBX is no contest. PBXs are very specialized and refined systems that are far more robust than any PC. VoIP through a Gateway This implementation is very similar to VoIP through a router, however, instead of using a router to route the calls; the functionality is part of the PBX. This can be a function of one of the cards in the PBX or simply a stand-alone device connected to the PBX. According to the Irwin Handbook of Telecommunications, Some manufacturers such as Lucent and Nortel provide IP trunk cards, but others do not, in which case the PBX would connect to either the router or the gateway through standard T1/E1 or analog tie trunk cards. Technology Essays.