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Voice Port Configuration Guide, Cisco IOS Release 12.4T
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Configuring Digital Voice Ports
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Contents
Configuring Digital Voice PortsLast Updated: December 14, 2011
The digital voice port commands discussed in this section configure channelized T1 or E1 connections; for information on ISDN connections, refer to the Cisco IOS ISDN Voice Configuration Guide. The T1 or E1 lines that connect a telephony network to the digital voice ports on a router or access server contain channels for voice calls; a T1 line contains 24 full-duplex channels or timeslots , and an E1 line contains 30. The signal on each channel is transmitted at 64 kbps, a standard known as Digital Signal 0 (DS0); the channels are known as DS0 channels. The ds0-group command creates a logical voice port (a DS0 group) from some or all of the DS0 channels, which allows you to address those channels easily, as a group, in voice-port configuration commands. Digital voice ports are found at the intersection of a packet voice network and a digital, circuit-switched telephone network. The digital voice port interfaces that connect the router or access server to T1 or E1 lines pass voice data and signaling between the packet network and the circuit-switched network. Signaling is the exchange of information about calls and connections between two ends of a communication path. For instance, signaling communicates to the call's endpoints whether a line is idle or busy, whether a device is on-hook or off-hook, and whether a connection is being attempted. An endpoint can be a central office (CO) switch, a PBX, a telephony device such as a telephone or fax machine, or a voice-equipped router acting as a gateway. There are two aspects to consider about signaling on digital lines: one aspect is the actual information about line and device states that is transmitted, and the second aspect is the method used to transmit the information on the digital lines. The actual information about line and device states is communicated over digital lines using signaling methods that emulate the methods used in analog circuit-switched networks: Foreign Exchange Service (FXS), Foreign Exchange Office (FXO), and Ear and Mouth (E&M). The method used to transmit the information describes the way that the emulated analog signaling is transmitted over digital lines, which may be common-channel signaling (CCS) or channel-associated signaling (CAS). CCS sends signaling information down a dedicated channel and CAS takes place within the voice channel itself. This chapter describes CAS, which is sometimes called robbed-bit signaling because user bandwidth is robbed by the network for signaling. A bit is taken from every sixth frame of voice data to communicate on- or off-hook status, wink, ground-start, dialed digits, and other information about the call. In addition to setting up and tearing down calls, CAS provides the receipt and capture of dialed number identification (DNIS) and automatic number identification (ANI) information, which are used to support authentication and other functions. The main disadvantage of CAS is its use of user bandwidth to perform these signaling functions. For signaling to pass between the packet network and the circuit-switched network, both networks must use the same type of signaling. The voice ports on Cisco routers and access servers can be configured to match the signaling of most COs and PBXs, as explained in this document. Finding Feature InformationYour software release may not support all the features documented in this module. For the latest feature information and caveats, see the release notes for your platform and software release. To find information about the features documented in this module, and to see a list of the releases in which each feature is supported, see the Feature Information Table at the end of this document. Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required. Prerequisites for Configuring Digital Voice PortsDigital T1 or E1 packet voice capability requires specific service, software, and hardware:
The memory required for high-volume applications may be greater than that listed. Support for digital T1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature set requires 8 MB of flash memory; other Plus feature sets require 16 MB.
For high-volume applications, the memory required may be greater than these minimum values. Support for digital E1 packet voice trunk network modules is included in Plus feature sets. The IP Plus feature set requires 16 MB of flash memory.
The memory required for high-volume applications may be greater than that listed. Support for T1/E1 high-capacity digital voice port adapters is included in Plus feature sets. The IP Plus feature set requires 16 MB of flash memory.
After the controllers have been configured, the show voice port summarycommand can be used to determine available voice port numbers. If the show voice port command and a specific port number is entered, the default voice-port configuration for that port displays. The following is show voice port summary sample output for a Cisco MC3810:
Router# show voice port summary
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
====== == ========== ===== ==== ======== ======== ==
0:17 18 fxo-ls down down idle on-hook y
0:18 19 fxo-ls up dorm idle on-hook y
0:19 20 fxo-ls up dorm idle on-hook y
0:20 21 fxo-ls up dorm idle on-hook y
0:21 22 fxo-ls up dorm idle on-hook y
0:22 23 fxo-ls up dorm idle on-hook y
0:23 24 e&m-imd up dorm idle idle y
Information About Digital Voice Hardware
Cisco 880 Series RoutersBeginning with Cisco IOS Release 12.4(15)XZ, the Cisco 880 series fixed router platforms support the implementation of analog (FXS/DID/FXO) and digital (BRI S/T) voice ports. The IAD881B, IAD881F, IAD888B, and IAD888F models support voice interface FXS or BRI. The IAD881F and IAD888F models have four FXS ports and the IAD881B and IAD888B models support two ports for ISDN BRI digital voice interface. In the IAD881B and IAD888B models, the voice BRI interface presents an ISDN S/T interface to connect either to an NT1 terminating an ISDN telephone network (TE-side) or to a TE user device such as an ISDN telephone or PBX (NT-side). In the IAD881B and IAD888B models, the BRI interface is available as the primary voice interface and is intended to be connected to a PBX (network side trunk). All the voice interfaces are onboard though they are recognized as a 4-port FXS VIC and a 2-port BRI VIC in order to leverage existing voice drivers. The C881and C888 SRST models automatically detect a failure occuring in the network and initiate a process to auto-configure the router. This process provides call-processing backup redundancy for the IP and FXS phones and helps to ensure that telephony capabilities stay operational. All the IP or analog phones hanging off of a telecommuter site are controlled by the headquarters office call control (Cisco Unified CallManager or CallManager Express). In case of a WAN failure, the telecommuter router allows all phones to re-register to it in SRST mode and allow all inbound and outbound dialing to be routed off to the PSTN (using back up FXO or BRI port). Upon restoration of WAN connectivity, the system automatically shifts call processing back to the primary Cisco Unified Call Manager cluster. Cisco 2600 Cisco 3600 and Cisco 3700 Series RoutersDigital voice hardware on Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series modular access routers includes the high-density voice (HDV) network module and the multiflex trunk (MFT) voice/WAN interface card (VWIC). When an HDV is used in conjunction with an MFT and packet voice DSP modules (PVDMs), the HDV module is also called a digital packet voice trunk network module. The digital T1 or E1 packet voice trunk network module supports T1 or E1 applications, including fractional use. The T1 version integrates a fully managed DSU/CSU, and the E1 version includes a fully managed DSU. The digital T1 or E1 packet voice trunk network module provides per-channel T1 or E1 data rates of 64 or 56 kbps for WAN services (Frame Relay or leased line). Digital T1 or E1 packet voice trunk network modules allow enterprises or service providers, using the voice-equipped routers as customer premises equipment (CPE), to deploy digital voice and fax relay. These network modules receive constant bit-rate telephony information over T1 or E1 interfaces and convert that information to a compressed format so that it can be sent over a packet network. The digital T1 or E1 packet voice trunk network modules can connect either to a PBX (or similar telephony device) or to a CO to provide PSTN connectivity. The MFT VWICs that are used in the packet voice trunk network modules are available in one- and two-port configurations for T1 and for E1, and in two-port configurations with drop-and-insert capability for T1 and E1. MFTs support the following kinds of traffic:
The digital T1 or E1 packet voice trunk network module contains five 72-pin Single In-line Memory Module (SIMM) sockets or banks, numbered 0 through 4, for PVDMs. Each socket can be filled with a single 72-pin PVDM, and there must be at least one packet voice data module (PVDM-12) in the network module to process voice calls. Each PVDM holds three DSPs, so with five PVDM slots populated, a total of 15 DSPs are provided. High-complexity codecs support two simultaneous calls on each DSP, and medium-complexity codecs support four calls on each DSP. A digital T1 or E1 packet voice trunk network module can support the following numbers of channels:
For more information, refer to the following:
Cisco 7200 and Cisco 7500 Series RoutersCisco 7200 and Cisco 7500 series routers support multimedia routing and bridging with a wide variety of protocols and media types. The Cisco 7000 family Versatile Interface Processor (VIP) is based on a reduced instruction set computing (RISC) engine optimized for I/O functions. To this engine are attached one or two port adapters or daughter boards, which provide the media-specific interfaces to the network. The network interfaces provide connections between the routers' peripheral component interconnect (PCI) buses and external networks. Port adapters can be placed in any available port adapter slot, in any desired combination. T1/E1 high-capacity digital voice port adapters for Cisco 7200 and Cisco 7500 series routers allow enterprises or service providers, using the equipped routers as CPE, to deploy digital voice and fax relay. These port adapters receive constant bit-rate telephony information over T1/E1 interfaces and can convert that information to a compressed format for transmission as VoIP. Two types of digital voice port adapters are supported on Cisco 7200 and Cisco 7500 series routers: two-port high-capacity (up to 48 or 120 channels of compressed voice, depending on codec choice), and two-port moderate capacity (up to 24 or 48 channels of compressed voice). These single-width port adapters incorporate two universal ports configurable for either T1 or E1 connection, for use with high-performance DSPs. Integrated CSU/DSUs, echo cancellation, and DS0 drop-and-insert functionality eliminate the need for external line termination devices and multiplexers. For more information, refer to the following publications:
Cisco AS5300The Cisco AS5300 includes three expansion slots. One slot is for either an Octal T1/E1/PRI feature card (eight ports) or a Quad T1/E1/PRI feature card (four ports), and the other two can be used for voice/fax or modem feature cards. Because a single voice/fax feature card (VFC) can support up to 48 (T1) or 60 (E1) voice calls, the Cisco AS5300 can support a total of 96 or 120 simultaneous voice calls. Cisco AS5300 VFCs are coprocessor cards, each with a powerful reduced instruction set computing (RISC) engine and dedicated, high-performance DSPs to ensure predictable, real-time voice processing. The design couples this coprocessor with direct access to the Cisco AS5300 routing engine for streamlined packet forwarding. For more information, refer to the following publications: Cisco AS5350 and Cisco AS5400 Universal GatewaysThe Cisco AS5350 and Cisco AS5400 universal gateways are versatile data and voice communications platforms that provide the functions of a gateway, router, and digital modems in a single modular chassis. The gateways are intended for Internet service providers (ISPs), telecommunications carriers, and other service providers that offer managed Internet connections, and also medium to large sites that provide both digital and analog access to users on an enterprise network. The cards that reside in the Cisco AS5350 and AS5400 chassis, sometimes referred to as dial feature cards (DFCs), are of two types: trunk cards, which provide an E1, T1, or T3 interface, and universal port cards, which host the universal DSPs that dynamically handle voice, dial, and fax calls. For more information, refer to the following publications: Cisco AS5800The Cisco AS5800 has two primary system components: the Cisco 5814 dial shelf (DS), which holds channelized trunk cards and connects to the PSTN, and the Cisco 7206 router shelf (RS), which holds port adapters and connects to the IP backbone. The dial shelf acts as the access concentrator by accepting and consolidating all types of remote traffic, including voice, dial-in analog and digital ISDN data, and industry-standard WAN and remote connection types. The dial shelf also contains controller cards voice feature cards, modem feature cards, trunk cards, and dial shelf interconnect cards. One or two dial shelf controllers (DSCs) provide clock and power control to the dial shelf cards. Each DSC contains a block of logic that is referred to as the common logic and system clocks. This block of logic can use a variety of sources to generate the system timing, including an E1 or T1/T3 input signal from the BNC connector on the front panel of the DSC. The configuration commands for the master clock specify the various clock sources and a priority for each source (see the Clock Sources on Digital T1 E1 Voice Ports). The Cisco AS5800 voice feature card is a multi-DSP coprocessing board and software package that adds VoIP capabilities to the Cisco AS5800 platform. The Cisco AS5800 voice feature card, when used with other cards such as LAN/WAN and modem cards, provides a gateway for up to 192 packetized voice/fax calls and 360 data calls per card. A Cisco AS5800 can support up to 1344 voice calls in split-dial-shelf configuration with two 7206VXR router shelves. For more information, refer to the following publications: Cisco AS5850 Universal GatewayThe Cisco AS5850 is a high-density ISDN and port WAN aggregation system that provides both digital and analog call termination. It is intended to be used in service-provider dial point-of-presence (POP) or centralized-enterprise dial environments. The feature cards and the route switch controller (RSC) communicate over a nonblocking interconnect that supports Fast Ethernet and full-duplex service. The Cisco AS5850 contains ingress interfaces (CT3 and CE1/PRI) that terminate ISDN and modem calls and break out individual calls (DS0s) from the appropriate telco services. Digital or ISDN calls are terminated on the trunk-card HDLC controllers, and analog calls are sent to port resources on the same card or on separate port cards. As a result, any DS0 can be mapped to any HDLC controller or port module. Unlike the Cisco AS5800, trunk-termination and port-handling services can be performed on the same card in the same slot. For more information, refer to the following publications: Cisco Catalyst 6500 Series Switches and Cisco 7600 Series RoutersThe Communication Media Module (CMM) acts as the VoIP gateway and media services module by using Media Gateway Control Protocol (MGCP), H.323, and SIP protocols with Cisco CallManager and other call agents. The CMM can support single or multiple Cisco CallManagers in an IP communication network. These VoIP gateway and media services features are provided through the four different types of CMM port adapters as shown in the table below.
For specific configuration information for the Catalyst 6500 series and Cisco 7600 series, see the following documents:
For specific installation and configuration information for the CMM, see the following document: Cisco MC3810To support a T1 or E1 digital voice interface, the Cisco MC3810 must be equipped with a digital voice interface card (DVM). The DVM interfaces with a digital PBX, channel bank, or video codec. It supports up to 24 channels of compressed digital voice at 8 kbps, or it can cross-connect channelized data from user equipment directly onto the router's trunk port for connection to a carrier network. The DVM is available with a balanced interface using an RJ-48 connector or with an unbalanced interface using BNC connectors. Optional HCMs can replace standard VCMs to operate according to the voice compression coding algorithm (codec) specified when the Cisco MC3810 is configured. The HCM2 provides 4 voice channels at high codec complexity and 8 channels at medium complexity. The HCM6 provides 12 voice channels at high complexity and 24 channels at medium complexity. You can install one or two HCMs in a Cisco MC3810, but an HCM cannot be combined with a VCM in the same chassis. For more information, refer to the following publications:
How to Configure Digital T1 E1 Voice PortsThis section describes commands for the basic configuration of digital voice ports. Make sure you have all the data recommended in the Prerequisites for Configuring Digital Voice Ports before starting these procedures. The basic steps for configuring digital voice ports are described in the next three sections. They are grouped by the configuration mode from which they are executed, as follows:
Configuring Codec Complexity on Digital T1 E1 Voice PortsThis section provides two configuration task tables: one for the Cisco 2600, Cisco 3600, and Cisco 3700 series routers and the Cisco MC3810 concentrator, which use voice-card configuration mode, and the second for the Cisco 7200 and Cisco 7500 series routers, which use DSP interface configuration mode. The task tables can be found in the following sections: Configuring Codec Complexity on Cisco 880 Series, Cisco 2600, Cisco 3600, Cisco 3700 Series and Cisco MC3810: Codec complexity refers to the amount of processing power assigned to a codec method on a voice port. On most router platforms that support codec complexity, codec complexity is selected in voice-card configuration mode, although it is selected in DSP interface mode on the Cisco 7200 and Cisco 7500 series. On the Cisco 880 series, Cisco 2600, Cisco 3600, Cisco 3700, Cisco 7200, and Cisco 7500 routers, codec complexity can be configured separately for each T1/E1 digital packet voice trunk network module or port adapter. On a Cisco MC3810, the codec complexity setting applies to both HCMs if two HCMs are installed.
To configure codec complexity for digital voice ports on the Cisco 880 series, Cisco 2600 series, Cisco 3600 series, and Cisco 3700 series routers, and for voice ports on HCMs on the Cisco MC3810, use the following commands: DETAILED STEPS
Changing Codec ComplexityTo change codec complexity on Cisco 880 Series, Cisco 2600 Series, Cisco 3600 Series, Cisco 3700 Series, and Cisco MC3810 after the controller and voice ports have already been configured, use the following commands:
DETAILED STEPS
Configuring the Flex Option on Codec ComplexityThe IP Communications High-Density Digital Voice/Fax Network Module feature enables the flex option for configuring codec complexity. On the Cisco 2600 XM, Cisco 2691, Cisco 3700 series routers, codec complexity can be configured using the flex option for configuring codec complexity. This option allows the DSP to process up to 16 channels. In addition to continuing support for configuring a fixed number of channels per DSP, the flex option enables the DSP to handle a flexible number of channels. The total number of supported channels varies from 6 to 16, depending on which codec is used for a call. Therefore, the channel density varies from 6 per DSP (high-complexity codec) to 16 per DSP (g.711 codec). The following requirements apply to the IP Communications High-Density Digital Voice/Fax Network Module feature.
Codec Combinations for DSP Sharing: When network modules or PVDM2s on the motherboard are configured for DSP sharing, the codec complexity has to match. A local resource sharing or importing from a remote network module must match its characteristics, that is, a high-complexity network module can only share from another high-complexity network module, whereas a flex-complexity network module can share DSPs from both high-complexity and flex-complexity network modules. The table below summarizes the codec combinations for DSP-sharing. Using Flex Mode In flex mode, you can connect (or configure in the case of DS0 groups and PRI groups) more voice channels to the module than the DSPs can accommodate. This is referred to as oversubscription. If all voice channels should go active simultaneously, the DSPs will be oversubscribed and calls that are unable to allocate a DSP resource will fail to connect.
To enable the IP Communications Voice/Fax Network Module feature, perform this task to configure the voice card for the flex option in codec complexity. DETAILED STEPS
Configuring Codec ComplexityOn Cisco 7200 series and Cisco 7500 series routers, codec complexity is configured in the DSP interface.
DETAILED STEPS
Cisco 7200 Series: On the Cisco 7200 series, the PA-MCX-2TE1 port adapter (PA) card can be used for making voice calls. This PA does not have any DSPs but uses the DSP resources of the PA-VXC-2TE1+ card present in another slot. If the PA-MCX card is used, codec complexity is configured for PA-VXC, while all other echo cancellation configurations are done for PA-MCX. The PA-MCX card borrows the DSP resources from the PA-VXC, PA-VXB, or PA-VXA card. If one of the PA-VXC, PA-VXB, or PA-VXA cards has extended echo cancellation configured on the DSP interface, extended echo cancellation is enabled for the PA-MCX card. It is recommended that you have the same codec complexity and echo cancellation configuration on all the PA-VXC, PA-VXB, or PA-VXA cards in the router. Cisco AS5300: Codec support on the Cisco AS5300 is determined by the capability list on the voice feature card, which defines the set of codecs that can be negotiated for a voice call. The capability list is created and populated when VCWare is unbundled and DSPWare is added to VFC flash memory. The capability list does not indicate codec preference; it simply reports the codecs that are available. The session application decides which codec to use. Codec support is configured on dial peers rather than on voice ports; refer to the "Dial Peer Configuration on Voice Gateway Routers" document. Cisco AS5800: Codec support is selected on Cisco AS5800 access servers during dial peer configuration. Refer to the "Dial Peer Configuration on Voice Gateway Routers" document. Configuring Controller Settings for Digital T1 E1 Voice PortsThe controller configuration for digital T1/E1 voice ports must match the line characteristics of the telephony network connection so that voice and signaling can be transferred between them and so that logical voice ports, or DS0 groups, may be established. Specific line characteristics must be configured to match those of the PSTN line that is being connected to the voice port. These are typically configured in controller configuration mode. The figure below shows how a ds0-group command gathers some of the DS0 time slots from a T1 line into a group that becomes a single logical voice port that can later be addressed as a single entity in voice port configurations. Other DS0 groups for voice can be created from the remaining time slots shown in the figure, or the time slots can be used for data or serial pass-through.
Voice port controller configuration includes setting the parameters described in the following sections: Another controller command that might be needed, cablelength, is discussed in the Cisco IOS Interface and Hardware Component Command Reference.
Framing Formats on Digital T1 E1 Voice PortsThe framing format parameter describes the way that bits are robbed from specific frames to be used for signaling purposes. The controller must be configured to use the same framing format as the line from the PBX or CO that connects to the voice port you are configuring. Digital T1 lines use SF or ESF framing formats. SF provides two-state, continuous supervision signaling, in which bit values of 0 are used to represent on-hook and bit values of 1 are used to represent off-hook. ESF robs four bits instead of two, yet has little impact on voice quality. ESF is required for 64-kbps operation on DS0 and is recommended for PRI configurations. E1 lines can be configured for CRC4 or no cyclic redundancy check, with an optional argument for E1 lines in Australia. Clock Sources on Digital T1 E1 Voice PortsDigital T1/E1 interfaces use timers called clocks to ensure that voice packets are delivered and assembled properly. All interfaces handling the same packets must be configured to use the same source of timing so that packets are not lost or delivered late. The timing source that is configured can be external (from the line) or internal to the router's digital interface. If the timing source is internal, timing derives from the onboard phase-lock loop (PLL) chip in the digital voice interface. If the timing source is line (external), then timing derives from the PBX or PSTN CO to which the voice port is connected. It is generally preferable to derive timing from the PSTN becauseits clocks are maintained at an extremely accurate level. This is the default setting for the clocks. When two or more controllers are configured, one should be designated as the primary clock source; it will drive the other controllers. The line keyword specifies that the clock source is derived from the active line rather than from the free-running internal clock. The following rules apply to clock sourcing on the controller ports:
This section describes the five basic timing scenarios that can occur when a digital voice port is connected to a PBX or CO. In all the examples that follow, the PSTN (or CO) and the PBX are interchangeable for purposes of providing or receiving clocking.
controller E1 1/0 framing crc4 linecoding hdb3 clock source internal ds0-group timeslots 1-15 type e&m-wink-start
The following configuration sets up this clocking method: controller T1 1/0 framing esf linecoding ami clock source line ds0-group timeslots 1-12 type e&m-wink-start
Because the PLL can derive clocking from only one source, this case is more complex than the two preceding examples. Before looking at the details, consider the following as they pertain to the clocking method:
In the dual voice ports receiving clocking from the line scenario, the PLL derives clocking from the CO and puts the voice port connected to the PBX into looped-time mode. This is usually the best method because the CO provides an excellent clock source (and the PLL usually requires that the CO provide that source) and a PBX usually must receive clocking from the other voice port. The following configuration sets up this clocking method (controller E1 1/0 is connected to the CO; controller E1 1/1 is connected to the PBX: controller E1 1/0 framing crc4 linecoding hdb3 clock source line primary ds0-group timeslots 1-15 type e&m-wink-start ! controller E1 1/1 framing crc4 linecoding hdb3 clock source line ds0-group timeslots 1-15 type e&m-wink-start The clock source line primary command tells the router to use this voice port to drive the PLL. All other voice ports configured as clock source line are then put into an implicit loop-timed mode. If the primary voice port fails or goes down, the other voice port instead receives the clock that drives the PLL. In this configuration, port 1/1 might see controlled slips, but these should not force it down. This method prevents the PBX from seeing slips.
The following configuration sets up this clocking method: controller E1 1/0 framing crc4 linecoding hdb3 clock source line ds0-group timeslots 1-15 type e&m-wink-start ! controller E1 1/1 framing crc4 linecoding hdb3 clock source internal ds0-group timeslots 1-15 type e&m-wink-start
The following configuration sets up this clocking method: controller E1 1/0 framing crc4 linecoding hdb3 clock source internal ds0-group timeslots 1-15 type e&m-wink-start ! controller E1 1/1 framing esf linecoding b8zs clock source internal ds0-group timeslots 1-15 type e&m-wink-start Network Clock TimingVoice systems that pass digitized (pulse code modulation or PCM) speech have always relied on the clocking signal being embedded in the received bit stream. This reliance allows connected devices to recover the clock signal from the bit stream, and then use this recovered clock signal to ensure that data on different channels keep the same timing relationship with other channels. If a common clock source is not used between devices, the binary values in the bit streams may be misinterpreted because the device samples the signal at the wrong moment. As an example, if the local timing of a receiving device is using a slightly shorter time period than the timing of the sending device, a string of eight continuous binary 1s may be interpreted as nine continuous 1s. If this data is then re-sent to further downstream devices that used varying timing references, the error could be compounded. By ensuring that each device in the network uses the same clocking signal, you can ensure the integrity of the traffic. If timing between devices is not maintained, a condition known as clock slip can occur. Clock slip is the repetition or deletion of a block of bits in a synchronous bit stream due to a discrepancy in the read and write rates at a buffer. Slips are caused by the inability of an equipment buffer store (or other mechanisms) to accommodate differences between the phases or frequencies of the incoming and outgoing signals in cases where the timing of the outgoing signal is not derived from that of the incoming signal. A T1 or E1 interface sends traffic inside repeating bit patterns called frames. Each frame is a fixed number of bits, allowing the device to see the start and end of a frame. The receiving device also knows exactly when to expect the end of a frame simply by counting the appropriate number of bits that have come in. Therefore, if the timing between the sending and receiving device is not the same, the receiving device may sample the bit stream at the wrong moment, resulting in an incorrect value being returned. Even though Cisco IOS software can be used to control the clocking on these platforms, the default clocking mode is effectively free running, meaning that the received clock signal from an interface is not connected to the backplane of the router and used for internal synchronization between the rest of the router and its interfaces. The router will use its internal clock source to pass traffic across the backplane and other interfaces. For data applications, this clocking generally does not present a problem as a packet is buffered in internal memory and is then copied to the transmit buffer of the destination interface. The reading and writing of packets to memory effectively removes the need for any clock synchronization between ports. Digital voice ports have a different issue. It would appear that unless otherwise configured, Cisco IOS software uses the backplane (or internal) clocking to control the reading and writing of data to the DSPs. If a PCM stream comes in on a digital voice port, it will be using the external clocking for the received bit stream. However, this bit stream will not necessarily be using the same reference as the router backplane, meaning the DSPs may misinterpret the data coming in from the controller. This clocking mismatch is seen on the router's E1 or T1 controller as a clock slip--the router is using its internal clock source to send the traffic out the interface but the traffic coming in to the interface is using a completely different clock reference. Eventually, the difference in the timing relationship between the transmit and receive signal becomes so great that the controller registers a slip in the received frame. To eliminate the problem, change the default clocking behavior through Cisco IOS configuration commands. It is absolutely critical to set up the clocking commands properly. Even though these commands are optional, we strongly recommend you enter them as part of your configuration to ensure proper network clock synchronization: network-clock-participate [slot slot-number | wic wic-slot | aim aim-slot-number network-clock-select priority{bri | t1 | e1} slot / port The network-clock-participate command allows the router to use the clock from the line via the specified slot/WIC/AIM and synchronize the onboard clock to the same reference. If multiple VWICS are installed, the commands must be repeated for each installed card. The system clocking can be confirmed using the show network clocks command.
Line Coding on Digital T1 E1 Voice PortsDigital T1/E1 interfaces require that line encoding be configured to match that of the PBX or CO that is being connected to the voice port. Line encoding defines the type of framing used on the line. T1 line encoding methods include AMI and B8ZS. AMI is used on older T1 circuits and references signal transitions with a binary 1, or "mark." B8ZS, a more reliable method, is more popular and is recommended for PRI configurations as well. B8ZS encodes a sequence of eight zeros in a unique binary sequence to detect line-coding violations. Supported E1 line encoding methods are AMI and HDB3, which is a form of zero-suppression line coding. DS0 Groups on Digital T1 E1 Voice PortsFor digital voice ports, a single command, ds0-group, performs the following functions: The numbering for the logical voice port created as a result of this command is controller:ds0-group-number , where controller is defined as the platform-specific address for a particular controller. On a Cisco 3640 router, for example, ds0-group 1 timeslots 1-24 type e&m-wink automatically creates the voice port 1/0:1 when issued in the configuration mode for controller 1/0. On a Cisco MC3810 universal concentrator, when you are in the configuration mode for controller 0, the ds0-group 1 timeslots 1-24 type e&m-winkcommand creates logical voice port 0:1. To map individual DS0s, define additional DS0 groups under the T1/E1 controller, specifying different time slots. Defining additional DS0 groups also creates individual DS0 voice ports. Most digital T1/E1 connections used for switch-to-switch (or switch-to-router) trunks are E&M connections, but FXS and FXO connections are also supported. These are normally used to provide emulated-OPX (Off-Premises eXtension) from a PBX to remote stations. FXO ports connect to FXS ports. The FXO or FXS connection between the router and switch (CO or PBX) must use matching signaling, or calls cannot connect properly. Either ground-start or loop-start signaling is appropriate for these connections. Ground-start provides better disconnect supervision to detect when a remote user has hung up the telephone, but ground-start is not available on all PBXs. Digital ground start differs from digital E&M because the A and B bits do not track each other as they do in digital E&M signaling (that is, A is not necessarily equal to B). When the CO delivers a call, it seizes a channel (goes off-hook) by setting the A bit to 0. The CO equipment also simulates ringing by toggling the B bit. The terminating equipment goes off-hook when it is ready to answer the call. Digits are usually not delivered for incoming calls. E&M connections can use one of three different signaling types to acknowledge on-hook and off-hook states: wink start, immediate-start, and delay-start. E&M wink start is usually preferred, but not all COs and PBXs can handle wink-start signaling. The E&M connection between the router and switch (CO or PBX) must match the CO or PBX E&M signaling type, or calls cannot be connected properly. E&M signaling is normally used for trunks. It is normally the only way that a CO switch can provide two-way dialing with DID. In all the E&M protocols, off-hook is indicated by A=B=1 and on-hook is indicated by A=B=0 (robbed-bit signaling). If dial pulse dialing is used, the A and B bits are pulsed to indicate the addressing digits. The are several further important subclasses of E&M robbed-bit signaling: In the original wink start handshaking protocol, the terminating side responds to an off-hook from the originating side with a short wink (transition from on-hook to off-hook and back again). This wink tells the originating side that the terminating side is ready to receive addressing digits. After receiving addressing digits, the terminating side then goes off-hook for the duration of the call. The originating endpoint maintains off-hook for the duration of the call. In Feature Group D wink-start with wink acknowledge handshaking protocol, the terminating side responds to an off-hook from the originating side with a short wink (transition from on-hook to off-hook and back again) just as in the original wink-start. This wink tells the originating side that the terminating side is ready to receive addressing digits. After receiving addressing digits, the terminating side provides another wink (called an acknowledgment wink ) that tells the originating side that the terminating side has received the dialed digits. The terminating side then goes off-hook to indicate connection. This last indication can be due to the ultimate called endpoint's having answered. The originating endpoint maintains an off-hook condition for the duration of the call. In the immediate-start protocol, the originating side does not wait for a wink before sending addressing information. After receiving addressing digits, the terminating side then goes off-hook for the duration of the call. The originating endpoint maintains off-hook for the duration of the call.
To configure controller settings for digital T1/E1 voice ports, use the following commands: DETAILED STEPS
Configuring Basic Voice Port Parameters for Digital T1 E1 Voice PortsFor FXO and FXS connections the default voice-port parameter values are often adequate. However, for E&M connections, it is important to match the characteristics of your PBX, so voice port parameters may need to be reconfigured from their defaults. Each voice port that you address in digital voice port configuration is one of the logical voice ports that you created with the ds0-group command. Companding (from compression and expansion), used in Step 6 of the following table, is the part of the PCM process in which analog signal values are logically rounded to discrete scale-step values on a nonlinear scale. The decimal step number is then coded in its binary equivalent prior to transmission. The process is reversed at the receiving terminal using the same nonlinear scale. Voice-port configuration mode allows many of the basic voice call attributes to be configured to match those of the PSTN or PBX connection being made on this voice port. In addition to the basic voice port parameters, there are commands that allow for the fine- tuning of the voice port configurations or for configuration of optional features. In most cases, the default values for these commands are sufficient for establishing voice port configurations. If it is necessary to change some of these parameters to improve voice quality or to match parameters in proprietary PBXs to which you are connecting, use the commands in the "Fine-Tuning Analog and Digital Voice Ports" section. After voice port configuration, make sure the ports are operational by following the steps described in these chapters: For more information on voice port commands, refer to the Cisco IOS Voice Command Reference
To configure basic parameters for digital T1/E1 voice ports, use the following commands:
DETAILED STEPS
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL: www.cisco.com/go/trademarks. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (1110R) Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental. © 2011 Cisco Systems, Inc. All rights reserved.
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