Showing posts with label IP Station Ports. Show all posts
Showing posts with label IP Station Ports. Show all posts

Friday

Remote Port Cabinet Options

Customers who want a single PBX system supporting communications requirements across multiple physical locations can use one of many systems offering a remote port cabinet/carrier option, in addition to systems similar to those described in the preceding paragraph. Even though the Ericsson, Alcatel, Cisco, and Sphere systems have a standard single system image distributed architecture capable of supporting station users dispersed across one or more customer premises, some PBXs can support only remote communications requirements with optional hardware. The first PBX system that supported a remote off-premises port cabinet option was the Northern Telecom SL-1. The SL-1 RPE option was introduced in 1982. Since the original SL-1 option most other leading PBX suppliers have introduced their own remote cabinet/carrier options. Current remote options differ greatly across system platforms. Remote port options may include:

  1. Multicarrier port cabinets designed to support up to 1,000 ports

  2. Single carrier cabinets designed to support about 300 ports

  3. Half-shelf carrier cabinets designed to support about 150 ports

  4. Single port interface circuit boards designed to support 24 or 36 ports

Customers with significant remote port requirements would benefit from a system that supports a remote equipment option minimizing the number of cabinets/carriers needed at the remote location, instead of con- figuring 10 or 20 remote interface boards, each with limited port capacities. However, a large remote equipment cabinet would be very expensive if the number of remote ports is limited. Growth requirements at the remote location must be taken into consideration, and the right remote equipment hardware should be selected at the initial installation.

There are several traditional design guideline issues for remote port configurations:

  1. Distance between the main and the remote locations

  2. Available transmission carrier facilities

  3. Traffic engineering requirements

  4. Available trunk access at remote location

  5. Survivability requirements

Most customer configurations, including a remote port cabinet/carrier, are within a local or regional geographic area. There are few, if any, remote configurations that are transcontinental. PBX remote options using PSTN T1-carrier transmission services can support customer configurations of several hundreds of miles between the main and remote locations. The Avaya Definity ECS remote option over copper T1-carrier is available at distances of up to 3,500 miles, depending on the interexchange carrier. Several PBX systems support remote cabinets with fiber optic cabling, without repeaters, at distances between 6 miles (Nortel Networks Meridian 1, Mitel SX-2000 Light) and 22 miles (Avaya Definity ECS). Fiber optic transmission is not commonly available from exchange carriers and is used mainly for campus settings or when right of way is available.

A benefit of using fiber optic cabling to support remote locations is the high bandwidth capacity of the link. Using a fiber optic link for the Definity or Meridian 1 systems provides the same number of traffic channels between the local TDM bus and the center stage switching complex as the link available for single premises configurations. Using a T1-carrier link limits traffic channels between the locations. Although a T1-carrier circuit using DS1 formatting can support up to 24 voice-grade channels, some of the channels used for the remote option must be reserved for control signaling. Most remote options can be configured with more than one T1-carrier circuit to provide sufficient communications channel capacity. For example, the Definity ECS can support between one and four T1-carrier circuits per remote EPN cabinet, to a maximum of 96 channels. In a maximum T1-carrier implementation, four of the channels are reserved for control signaling, leaving 92 two- way channels for intercabinet voice communications. The dispersed switch network design of the Definity ECS G3r model supports local switching at the remote location, thereby minimizing the number of required center stage switch connections at the main location. The Meridian 1 can support up to three T1-carrier circuits for each remote IPEM cabinet, with 22 available one-way channels per link for all remote location call connections. The centralized switch network design of the intermediate and large Meridian 1 options requires that all remote calls be connected through the center stage switching complex at the main location, thus placing a significant traffic burden on the limited T1-carrier spans. The Siemens Hicom 300H remote communications module (RCM) option is supported with only one or two T1-carrier spans: two channels per span are reserved for control signaling, and the remaining 22 communications channels are one way. Like the Meridian 1, the Hicom 300H centralized switching design requires all remote calls be connected through the main location center stage switch.

The limitations of T1-carrier channel capacity require that traffic engineering support acceptable QoS standards. The number of available communications channels, one- or two-way, will depend on the number of remote location ports. The Definity EPN cabinet can easily support more than 1,000 ports (stations and trunks) at moderate traffic capacity, whereas Avaya usually recommends an 800 station limit. Customers must carefully analyze traffic flow to decide whether one, two, three, or four T1-carrier circuits are needed to support traffic between the remote EPN location and the main or other remote locations. The Nortel and Siemens solutions support fewer remote port capacities per cabinet carrier, but there are far fewer communications channels, and all remote call connections, including direct trunk access at the remote location, require two one-way channels for center stage switch connections.

It is desirable for station users at the remote location to have direct access to local exchange carrier trunk circuits terminated at the remote cabinet/carrier. There are many instances when local area calls placed by station users at the remote customer location would become long distance calls if routed to trunk circuits at the main PBX system. In contrast, it is usually preferable for PSTN long distance or private network calls placed at the remote location to route through the main location, where most trunks are concentrated and trunk access is more highly engineered.

If the control signaling link between the main and remote locations is down or there are transmission errors, the call processing and switching functions at the remote site will be affected, unless there are redundant design elements. The most basic redundant design option is duplication of the T1-carrier or optical fiber links connecting the main and remote locations. Almost all systems with remote port options are available with this option, although some systems have a greater degree of redundancy than others. For example, each of the T1-carrier links supporting the remote Definity EPN cabinets can be fully duplicated; the Meridian 1 design supports two active T1-carrier links and a spare T1-carrier link for emergency backup.

If redundant links are not an option or not configured, another redundant capability is local call control processing and switching capabilities at the remote location. Remote location survivability is an important issue for most customers. For example, there are many customers who support distributed call center operations across locations on a single system platform. Loss of an ACD agent group for a few minutes or hours could translate into significant revenue losses, in addition to reduced customer service levels. If a remote location is strategic to the enterprise operation, available redundant processing must be considered. Intecom, the first company to design a fully distributed PBX cabinet design, also introduced the first survivable remote cabinet option in the mid-1980s. Several other system suppliers have since shipped local call control processing options for their remote port cabinets for emergency situations when the control link to the main common control complex is not available.

Manufacturers of traditional circuit switched PBX systems have also taken advantage of ToIP technology by offering remote carrier options supported by IP-based transmission links. Avaya’s R300 Remote Office Communicator and Nortel’s i9150 remote options are designed to provide communications support for remote branch offices working behind a larger main location with a centralized PBX system. The R300, based on an Ascend Communications remote access concentrator, at the remote location can support analog, digital, and IP telephones and digital and analog local trunk circuit connections. The R300 is supported over an IP link supported at the main PBX location by Definity IP Media Processing and CLAN circuit boards. The Nortel remote cabinet shelf supports standard Meridian 1 peripherals and has local switching capability. A Meridian 1 supports the IP-connected i9150 using an integrated telephony gateway circuit board. A major benefit of the i9150 is that it has an integrated processor board providing backup call processing should there be link failure to the main location. The Avaya and Nortel solutions are targeted at customers with existing LAN/WAN infrastructures across customer premises locations.

Monday

IP Station Ports | Converged IP-PBX System Design

The key word in the above design attributes is integrated. A circuit switched PBX that supports IP peripherals using external gateway equipment is not considered an IP-PBX. A PBX system must be configured with integrated signaling or port interface circuitry to support ToIP communications if it is classified as an IP-PBX; otherwise any older PBX system with a T1/E1 interface and a third-party VoIP gateway could be considered an IP-PBX.

Most first-generation converged IP-PBX systems were based on a traditional proprietary common control complex. The common control complex, specifically the main system processor, functions as the gatekeeper for IP clients. A gatekeeper was originally defined as a component in an H.323 communications system used for admission control and address resolution. Gatekeepers allow calls to be placed directly between two IP endpoints with the use of peer-to-peer LAN switched connections; two IP telephones can communicate over a LAN/WAN infrastructure without using the PBX’s internal circuit switching network. A converged IP-PBX system also supports calls between an IP endpoint and a non-IP endpoint using gateway resources. The primary functions of a traditional gatekeeper are address translation, admissions control, bandwidth management, and zone control. The IP-PBX main processing unit assumes this responsibility for all IP communications. Most converged IP-PBXs support H.323 standards for LAN-based real-time multipoint communications networks.

A traditional circuit switched Meridian 1 Option 81C model, with its traditional core module cabinet can be upgraded to a converged IP-PBX system platform through the simple addition of an optional ITG line card and a software upgrade capable of supporting IP telephony. The optional interface card supports IP telephones by providing physical connectivity to the LAN, which links IP-PBX common equipment to the desktop IP voice terminal for control and voice communications signaling. The ITG line interface card converts on-board gateway resources for conversion between PCM and IP packet signals connectivity to an Ethernet LAN through a 10BaseT or 10/100BaseT interface, and passes gatekeeper control signals between the PBX call processor and IP endpoints.

In some system designs the IP port card with gateway resources also functions as a proxy server for the system gatekeeper to transmit control signals to and from IP endpoints. Nortel’s ITG line card is used for TCP/IP control signaling connections between the PBX system and the LAN. IP line interface cards from Siemens and Alcatel also support the server as proxy gatekeeper servers. Gatekeeper signaling also may be transmitted with a dedicated Ethernet TCP/IP interface circuit card, such as Avaya’s Definity/IP 600 CLAN interface card, or through an Ethernet connector located within the common control complex, such as a daughterboard on the CPU board (NEC’s NEAX2000 IPS). The terms gatekeeper and gateway were originally defined for H.323 LAN-based telephony systems but are now also used to describe the role of the PBX common control complex and new IP port interface cards used by circuit switched PBXs to support IP endpoints and connections. Although IP port interface cards might not support gatekeeper signaling, they all support gateway functions.

A gateway is composed of two elements: an MGC and an MG. The MGC handles call signaling and other non–media-related functions. The MG handles the communications media transmission. H.323 gateway functions typically include protocol conversion, connectivity, compression, decompression, fax demodulation, and fax remodulation. The latter two gateway functions are not commonly used with converged IP-PBX IP port interface cards because fax terminals are supported with traditional analog line interface circuit cards. The PBX gateway function is extremely important because TDM/PCM ports are likely to remain a significant percentage of the total number of converged IP-PBX peripherals endpoints for several years to come. Communications between IP ports and non-IP ports require protocol conversion between the different voice coding formats and connectivity to the local TDM bus via the gateway bearer channels.

The IP port card gateway resources can also be used to establish voice communications links between IP endpoints using different voice codecs. For example, an IP endpoint using G.729/A format for voice packeting cannot communicate with an IP endpoint using only a G.711 format unless there is a conversion process between the endpoints. IP port card gateway resources can perform this conversion process, which is known as transcoding. Transcoded IP calls may require circuit switched connections on a local TDM bus, unless the IP port card can redirect the reformatted communications signals back across the LAN to the IP endpoints.

IP line card capabilities for a converged IP-PBX differ across manufacturers. There are several parameters that can differentiate IP port interface capabilities:

  • Maximum number of IP stations supported (based on gatekeeper signaling)

  • Gateway channel resources

  • Voice codecs supported

The number of gateway channel connections per card is based on the number of on-board DSP resources, referred to by some system manufacturers as digital compressors. The term compressor signifies the compression algorithm function that converts the 64-Kbps PCM transmission format to the lower transmission rates supported by different IP voice codecs, such as those conforming to G.729 formatting standards (8 Kbps). PCM transmission may or may not be compressed to conserve LAN/WAN bandwidth resources. IP station transmission across the gateway to the local TDM bus needs to be decompressed (if not using the uncompressed, 64-Kbps G.711 format standard) to communicate with non-IP stations or access PSTN trunk circuits. The most common IP voice codecs supported by IP port interface cards are G.711, G.723.1, and G.729/A. The specific voice codec used for each IP call can be predefined by the system administrator, based on the call type (on-premises, off-premises) or specific number dialed.

There are different approaches used by converged IP-PBX system manufacturers in their IP station interface card designs. One approach is to limit the gateway channel capacity to support physical IP stations and provide nonblocking access to the local TDM bus in support of IP to non-IP calls. Siemens employs this strategy by limiting IP port capacity, based on gatekeeper signaling capacity, to 30 stations per interface card, the same as the maximum number of gateway channel links to the local TDM bus. The 1:1 ratio between IP station and gateway connections to the local TDM bus eliminates the possibility of blocked communications signaling to and from the TDM bus and LAN. Ericsson also implements this design strategy on its MD-110 for its IP station card.

Other leading converged IP-PBX system suppliers take a different approach. The Nortel Networks ITG line card supports 96 IP telephones, but only 24 gateway channels to the local TDM bus. The Alcatel OmniPCX 4400 INT-IP line interface card can support up to 500 IP clients, but the maximum number of gateway connections to the local TDM bus is limited to 60 gateway channels. The actual number of gateway channels per INT-IP card can be configured based on the number and type of daughterboards (8 or 30 compressors). An Alcatel system is unlikely to be configured with 500 IP Reflexes telephones supported by a single INT-IP board because of the extreme ratio of potential IP telephones to local TDM channel connections, unless there are minimal requirements for IP port to non-IP port connections. The reason Alcatel designed their interface card with a limited number of channels for a large number of physically connected telephones, and other manufacturers such as Nortel Networks designed IP station interface cards with potential contention for limited channels, is that gateway channels serve as pooled resources for any IP endpoint and can be configured based on traffic engineering requirements. For all the systems under discussion, IP interface cards are not dedicated to specific endpoints for gatekeeper or gateway operations. IP telephone communications services can be performed by any of the IP port interface cards in the system, a major difference from the support of traditional analog and digital telephones by dedicated port interface cards.

NEC and Avaya have designed IP port interface cards that handle only gateway functions because gatekeeper signaling is transmitted to IP endpoints through other means. The Avaya IP Media Processor Interface, sometimes referred to as a prowler board, supports between 32 and 64 channel connections, based on the IP voice codec implemented per call, G.711 or G.729/A. The Avaya port interface card has a total of 64 DSP resources. Each DSP resource can support a single G.711 communications interface, for a maximum total of 64 simultaneous conversations (1 DSP resource = 1 active gateway channel). The G.729/A protocol format uses compressed voice (8 Kbps) and requires two DSP resources per IP call requiring a circuit switched connection to a non-IP peripheral, for a maximum of 32 simultaneous conversations. The conversion process for G.729/A is more processing intensive than uncompressed G.711 (64 Kbps). The actual number of available channel connections to the Avaya Definity local TDM bus will change continually based on the type of voice codecs that are implemented for concurrent IP calls requiring TDM bus connectivity. The number of IP Media Processor Interface cards required in a system design will depend on IP call requirements for TDM bus connectivity, or transcoding. The NEC IP line interface card, known as the IP PAD, has 32 on-board compressors that perform the gateway function for IP stations requiring circuit switched connections to non-IP ports. NEC designed the card to have a maximum of 32 compressors because it occupies a single card slot that is limited to 32 TDM bus backplane connections.

Taking into account gatekeeper signaling and gateway resource capacities, the actual number of equipped and active IP endpoints that can be supported by an IP port interface card to maintain an acceptable QoS level is based on the number of IP endpoints, customer traffic patterns, and engineering requirements. The more likely a gateway channel will be used per IP station call, the lower the acceptable ratio between peripheral endpoints and channels. For example, if there are very few IP stations equipped in a converged IP-PBX and most ports are traditional TDM/PCM peripherals, a call generated by an IP telephone more likely will require a channel for connectivity to the local TDM bus to talk with a non-IP station/trunk. If a converged IP-PBX system is equipped with a very high percentage of IP telephones, as compared with traditional analog or digital telephones, and a significant percent- age of off-premises calls are handled over an IP WAN, the acceptable ratio between IP endpoints and gateway channels will be larger. A converged IP-PBX system design using separate interfaces for gatekeeper signaling transmission and gateway functions theoretically can be equipped with no gateway interface cards, assuming there is no requirement for any IP call to use TDM bus resources. This scenario is almost impossible to envision in a current customer environment, unless it is a small satellite location with all IP stations and no local central office trunking is IP networked to a main PBX system.

IP port interface circuit cards typically support functions beyond their gateway responsibilities. Some of these functions are administration and maintenance support, echo cancellation, silence suppression, DTMF detection, conferencing, and improved voice quality through dynamic jitter buffers. Basic administration/maintenance functions include support of station registration and initialization, downloading firmware changes to desktop terminals, and diagnostics of errors and alarm conditions at the port endpoint. Echo cancellation technology reduces the effects of echo heard by a caller when on an active voice call. If the echo transmission signal is delayed, the resulting echo will be noticeable to the caller. An echo canceller monitors caller speech; when an echo occurs, a signal is generated, transmitted, and sent back to the caller to cancel the echo. Silence suppression is the detection of the absence of audio on the bearer communications channels. Another term for silence suppression is voice activation detection. When no voice packets are detected, the gateway bearer communications channels are released from a call and made available for voice packets transmitted by another caller.

The jitter buffering function is very important for maintaining voice QoS levels. The time for voice packets to be transmitted and received between endpoints is known as delay. The “end-to-end” delay time consists of two network delay elements, fixed and variable. Jitter is the difference between the two delay values of two voice packets in the transmission across the network. Fixed network delay may include propagation delay of signals between the two endpoints, voice encoding delay, and the voice packeting time for the IP voice codec. Fixed network delay times can be calculated and corrected for, but variable delay times present a different problem. Variable packet delays can be caused by network traffic congestion and serialization delays on network interfaces. The quality of voice communications is degraded if there is a variation in the arrival of voice packets at the receiving endpoint. Network congestion can occur any time and cause variations in arrival times. To compensate for delay variations, the IP station card equipped with jitter buffers turn delay variations into a constant value so that voice can be transmitted and played out smoothly. Digital signal processing (DSP) algorithms can be programmed for different buffer times based on the expected voice packet network delay. The algorithm can review each packet’s timestamp included in the RTP header of the voice packet, calculate expected delays, and adjust the jitter buffer size accordingly. Extra buffer time can be programmed to account for variable packet delays and smooth out the packet flow. If packet delay exceeds the total jitter buffer time, the packet is dropped. Loss of one packet in a voice call does not significantly affect the quality of the call. If variable delay of voice packets is underestimated and many packets are dropped, the resulting voice call quality will suffer.

IP station equipment supported by an IP station interface card may include an IP telephone terminal instrument or a PC client softphone. Converged IP-PBX systems can support local IP stations that are physically located on the customer premises or remote at off-premises locations. Control signaling to and from the IP endpoint is over LAN/WAN facilities. A remote IP station can use private line WAN carrier facilities or PSTN services (ISDN, DSL, dial-up) for gatekeeper control signaling. A centrally housed IP port circuit card provides gateway function to non-IP ports.

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