BSc Systems 3 Unified Communications Chapter 1

Transcription

1 Chapter 1 - Cisco Voice Gateways Voice gateways are an integral part of the Voice over Internet Protocol (VoIP) network. In their most simple form, gateways are used to connect dissimilar networks, such as the IP network to the traditional Public Switched Telephone Network (PSTN). In essence, the primary function of a gateway is to convert data travelling through it to a format that the other side understands. The CVOICE exam objectives covered in this chapter are: Describe the function of gateways Describe DSP functionality Describe the different types of voice ports and their usage Describe dial peer types Describe codec s and codec complexity This chapter is divided into the following sections Voice Terminology Voice Gateways DSP Resources Circuit Signalling Analogue Circuits FXO and FXS Ports E&M Ports Digital Circuits Integrated Services Digital Network (ISDN) Voice Dial Peers Voice Encoding Voice Terminology Before delving into detail on gateways, it is imperative to have an understanding of the jargon used in traditional voice and VoIP implementations. This section does not provide significant detail on the phrases listed below, nor does it list all terminology currently used in voice networking. Instead, it provides a basic definition of some of the most common the terms that are used in this chapter, as well as in this guide. Keep in mind, however, that some of these terms will be expanded upon in this chapter and throughout this guide. The following is a list of terms and acronyms that are generally used in voice networking: ANI Bandwidth Consumption

2 CAC Codec Caller ID CLID CNID Delay DNIS Echo Jitter Packetization PSTN Serialization TDM VAD VoIP ANI Automatic Number Identification (ANI) refers to the telephone number of the calling party. Bandwidth Consumption Bandwidth consumption is the characteristic bandwidth required between two points to transport traffic. In other words, it is the amount of bandwidth required to transport data from point A to point B, for example. In voice networking, bandwidth is moderated by several methods of Call Admission Control (CAC). CAC Call Admission Control (CAC) is the tracking of bandwidth resources and traffic requests for the purposes of preventing oversubscription within the constraints of the current bandwidth. CAC prevents endpoints from sending more information, or attempting to send more information, than the available bandwidth can handle. Codec A codec, or coder / decoder, is a means of sampling, then digitally representing voice with a given level of compression. Caller ID

3 Caller ID (CID) describes the delivery of a calling name or number, or both. CLID Calling Line Identifier (CLID) is similar to ANI and refers to the number of the calling party. CNID The Calling Number Identification or Calling Name Identification is the same as the ANI. Delay Delay is the latency -- one way or round-trip -- that is incurred when packets are transmitted from one location to another. Delay includes the amount of processing time for intervening devices between source and destination. There are two types of delay: fixed and variable delay. DNIS The Dialed Number Identification Service is the telephone number of the called party. This refers to the number that the telephone company sends to the destination. The DNIS may or may not be the actual number the calling party dials to reach the destination. For example, the caller may dial one number, but it may be translated or redirected by the telephone provider and those digits would be sent to the destination. Echo Echo is the reflection of sound caused by a mismatch in impedance. Impedance measures how easily a circuit conducts current when a voltage runs through it. Impedance is a way of telling you how much of the voltage introduced at one end will really make it to the other end. Impedance is measured in ohms. Jitter Jitter is the variation in delay incurred in a given voice or packet stream. The varying arrival time of the packets can cause gaps in the recreation and playback of the original voice signal, which is undesirable and annoying to the listener. Jitter can be addressed by using de-jitter buffers. Packetization Packetization is the time it takes to put digital voice information into packets and remove the information from packets. Packetization delay is the time taken to fill a packet payload with encoded or compressed speech. This delay is a function of the sample block size required by the voice encoder (vocoder) and the number of blocks placed in a single frame. Packetization delay can also be called Accumulation delay, as the voice samples accumulate in a buffer before they are released. As a general rule, strive for a packetization delay of no more than 30ms. PSTN The Public Switched Telephone Network (PSTN) is the familiar telephone network, comprising of a Central Office (CO) and tie lines, which provide local and long distance calling. The PSTN is largely governed by technical standards created by the ITU-T, and uses E.164 addresses, which are telephone numbers. E.164 addresses will be described in detail later in this chapter. On the PST, the local telephone provider services neighborhoods via a local CO. The CO is the place where the telephone company terminates lines and switches calls between different locations. On the network side, the CO is typically connected to a regional CO that connects to the digital WAN, allowing the local telephone provider to provide local, national long distance, and even international long distance service to subscribers.

4 On the customer facing side, the telephone company may employ the use of remote switching facilities between the subscriber premise and the CO. While most signals received from the subscriber premise are analog signals, some remote switching facilities have the ability to convert those received analog signals into digital signals before forwarding them on to the local CO. The portion of the PSTN that connects the home to the local CO is referred to as the local loop or last mile. Going into detail on the PSTN is beyond the scope of the current CVOICE. Serialization Serialization is the insertion of bits onto a link. Serialization delay is the amount of time it takes for a networking device, such as a router, to encode a packet onto the wire for transmission. Serialization delay is incurred when encapsulating or segmenting a data stream into packets for egress from a given interface. The interface must service the packets one at a time, which in turn results in the delay. TDM Multiplexing is a form of transmission that allows multiple signals to travel simultaneously over a single medium. Time Division Multiplexing (TDM) is a means of providing access to transmission media though the use of a time slot. Several signals may be interleaved on the transmission media through the use of multiple time slots. The greater the number of time slots, the shorter the duration of each. VAD Voice Activity Detection (VAD) is a voice encoding algorithm that takes note of silence during voice conversations and suppresses the transmission of voice packets that contain no actual data within them. VoIP Voice over Internet Protocol (VoIP) is a means of compressing voice using a standardized codec, then encapsulating the results within IP for transport over data networks.

5 Voice Gateways Voice gateways are devices that communicate with other voice gateways, gatekeepers, their respective endpoints, and call control agents, such as Cisco Unified Communications Manager or a PBX via voice signaling and media protocols which include MGCP, H.323, SIP, SCCP and RTP. Voice signaling and media protocols will all be described in detail later in this guide. Function of Voice Gateways In VoIP networks, the primary purpose of voice gateways is to provide an interface between the VoIP network and the Public Switched Telephone Network (PSTN). This traditional use of voice gateways is illustrated in the following diagram: As illustrated in the diagram above, the voice gateway is connected to both the VoIP network and the PSTN. The gateway interfaces with the IP network and the PSTN and supports IP signaling control protocols used in Voice over IP, and Time Division Multiplexing (TDM) control protocols used on the PSTN. TDM will be described in detail later in this chapter. Voice gateways are also used to connect analog phones, fax machines and modems over IP networks as illustrated in the following diagram: In hybrid environments that include IP PBX solutions, such as Cisco Unified Communications Manager, and legacy TDM PBX solutions, voice gateways can be used to facilitate communication between the endpoints connected to the IP PBX and the TDM PBX, thus merging the two environments as illustrated in the following diagram:

6 When integrated into the network, the voice gateway allows for communication between the different network types and performs the following tasks, amongst many others: Interfaces with the IP network and the PSTN or PBX Performs call setup and teardown for calls between the VoIP and PSTN Relays Dual Tone Multi Frequency (DTMF) tones Supports IP and TDM control protocols Supports analog fax machines and modems over the IP network Supports call survivability when no Cisco Unified Communications Manager is available Supports Cisco Unified Communications Manager redundancy in CUCM networks Provides supplementary services such as hold, transfer, and conferencing Voice Gateway Components In their most basic form, voice gateways contain an IP interface and a legacy telephone interface. This allows the voice gateway to allow for communication between the IP network and the telephone network by translating between the different transmission formats and protocols used on the two networks. Cisco voice gateways support analog or digital telephony ports, and sometimes both port types, in addition to IP interfaces. Supported analog ports include the following: Foreign Exchange Office Ear and Mouth

7 Foreign Exchange Station Foreign Exchange Office (FXO) ports are used to connect to the PSTN or to a PBX. These are the ports on subscriber devices, such as analog telephones, fax machines and modems that connect to the PSTN via an FXS port. An Ear and Mouth (E&M) port is used to interconnect PBX devices using dedicated circuits from the PSTN. A Foreign Exchange Station (FXS) port is used to connect analog devices, such as analog telephones, fax machines, and modems. FXS ports are the ports that subscribers plug a telephone, fax machine or modem into. FXS ports provide telephony services for these devices. FXO, E&M, and FXS ports will be described in detail later in this chapter in the section pertaining to voice ports. Cisco voice gateways also support digital ports and interfaces, which include the following: T1 Primary Rate Interface T1 Channel Associated Signaling E1 Primary Rate Interface E1 R2 ISDN Basic Rate Interface Integrated Services Digital Network (ISDN) T1 Primary Rate Interface (PRI) provides 23 voicebearer channels, while ISDN E1 PRI provides 30 voice-bearer channels. ISDN T1 PRI standard is used in the United States, while ISDN E1 PRI is used in Europe. T1 Channel Associated Signaling (CAS), also referred to as robbed-bit signaling, is performed on voice channels by robbing the Least Significant Bit (LSB) of information in the channel. CAS will be described in detail later in this chapter. E1 R2 is a Channel Associated Signaling (CAS) system that is used in Europe. And finally, ISDN Basic Rate Interface (BRI) provides two voice-bearer channels for data and voice traffic, in addition to a single data channel for signaling traffic. As is the case with analog ports and circuits, digital ports and circuits will be described in detail later in this chapter on voice ports. The following output shows a Cisco 3845 ISR voice gateway with two GigabitEthernet interfaces, two E1-R2 digital ports, and eight FXO analog ports: R1#show version Cisco IOS Software, 3800 Software (C3845-ADVENTERPRISEK9_IVS-M), Version 12.4(18b), RELEASE SOFTWARE (fc2) Technical Support: Copyright (c) by Cisco Systems, Inc. Compiled Mon 19-May-08 21:58 by prod_rel_team ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1)

8 R uptime is 2 weeks, 6 days, 3 hours, 9 minutes System returned to ROM by reload at 20:31:52 UTC Tue Oct System image file is "flash:c3845-adventerprisek9_ivs-mz b.bin" This product contains cryptographic features and is subject to United States and local country laws governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute or use encryption. Importers, exporters, distributors and users are responsible for compliance with U.S. and local country laws. By using this product you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately. A summary of U.S. laws governing Cisco cryptographic products may be found at: If you require further assistance please contact us by sending to export@cisco.com. Cisco 3845 (revision 1.0) with K/46080K bytes of memory. Processor board ID FTX1306A09R 2 Gigabit Ethernet interfaces 2 Channelized E1/PRI ports 1 Virtual Private Network (VPN) Module 8 Voice FXO interfaces DRAM configuration is 64 bits wide with parity enabled. 479K bytes of NVRAM.

9 255488K bytes of ATA System CompactFlash (Read/Write) Configuration register is 0x2102 The voice ports on the voice gateway can be viewed by issuing the show voice port summary command, as illustrated in the following output on the same router illustrated above: R1#show voice port summary PORT CH SIG-TYPE ADMIN OPER IN STATUS OUT STATUS EC 0/0/0 -- fxo-ls up dorm idle on-hook y 0/0/1 -- fxo-ls up dorm idle on-hook y 0/0/2 -- fxo-ls up dorm idle on-hook y 0/0/3 -- fxo-ls up dorm idle on-hook y 0/1/0 -- fxo-ls up dorm idle on-hook y 0/1/1 -- fxo-ls up dorm idle on-hook y 0/1/2 -- fxo-ls up dorm idle on-hook y 0/1/3 -- fxo-ls up dorm idle on-hook y 1/0:1 01 r2-digital up dorm idle idle y 1/0:1 02 r2-digital up dorm idle idle y 1/0:1 03 r2-digital up dorm idle idle y 1/0:1 04 r2-digital up dorm idle idle y 1/0:1 05 r2-digital up dorm idle idle y 1/0:1 06 r2-digital up dorm idle idle y 1/0:1 07 r2-digital up dorm idle idle y 1/0:1 08 r2-digital up dorm idle idle y 1/0:1 09 r2-digital up up answered idle y 1/0:1 10 r2-digital up dorm idle idle y 1/0:1 11 r2-digital up dorm idle idle y 1/0:1 12 r2-digital up dorm idle idle y 1/0:1 13 r2-digital up dorm idle idle y 1/0:1 14 r2-digital up dorm idle idle y 1/0:1 15 r2-digital up dorm idle idle y 1/0:1 17 r2-digital up dorm idle idle y 1/0:1 18 r2-digital up dorm idle idle y 1/0:1 19 r2-digital up up answered idle y 1/0:1 20 r2-digital up dorm idle idle y 1/0:1 21 r2-digital up dorm idle idle y 1/0:1 22 r2-digital up dorm idle idle y 1/0:1 23 r2-digital up dorm idle idle y 1/0:1 24 r2-digital up up answered idle y 1/0:1 25 r2-digital up dorm idle idle y 1/0:1 26 r2-digital up dorm idle idle y 1/0:1 27 r2-digital up dorm idle idle y 1/0:1 28 r2-digital up up answered idle y 1/0:1 29 r2-digital up dorm idle idle y 1/0:1 30 r2-digital up dorm idle idle y 2/0:1 01 r2-digital up dorm idle idle y

10 2/0:1 02 r2-digital up dorm idle idle y 2/0:1 03 r2-digital up dorm idle idle y 2/0:1 04 r2-digital up dorm idle idle y 2/0:1 05 r2-digital up dorm idle idle y 2/0:1 06 r2-digital up dorm idle idle y 2/0:1 07 r2-digital up dorm idle idle y 2/0:1 08 r2-digital up dorm idle idle y 2/0:1 09 r2-digital up dorm idle idle y 2/0:1 10 r2-digital up dorm idle idle y 2/0:1 11 r2-digital up dorm idle idle y 2/0:1 12 r2-digital up dorm idle idle y 2/0:1 13 r2-digital up dorm idle idle y 2/0:1 14 r2-digital up dorm idle idle y 2/0:1 15 r2-digital up up answered idle y 2/0:1 17 r2-digital up dorm idle idle y 2/0:1 18 r2-digital up dorm idle idle y 2/0:1 19 r2-digital up dorm idle idle y 2/0:1 20 r2-digital up dorm idle idle y 2/0:1 21 r2-digital up dorm idle idle y 2/0:1 22 r2-digital up dorm idle idle y 2/0:1 23 r2-digital up dorm idle idle y 2/0:1 24 r2-digital up dorm idle idle y 2/0:1 25 r2-digital up dorm idle idle y 2/0:1 26 r2-digital up dorm idle idle y 2/0:1 27 r2-digital up dorm idle idle y 2/0:1 28 r2-digital up dorm idle idle y 2/0:1 29 r2-digital up dorm idle idle y 2/0:1 30 r2-digital up dorm idle idle y Voice Gateway Capabilities When choosing a voice gateway for Cisco VoIP networks, it is important to ensure that the selected gateway supports the following four core requirements: 1. Dual Tone Multi Frequency Relay 2. Supplementary Services 3. Communications Manager Redundancy 4. Call Survivability DMTF, or Dual Tone Multi Frequency, refers to the tones that are played when the dial pad on a phone is pressed. DTMF is often referred to as touch-tones. Because voice traffic is typically compressed, DTMF can become distorted. The DTMF relay feature allows the DTMF to be sent outof-band, which resolves the distortion problem. The term out-of-band refers to communications which occur outside of a previously established communication method or channel. In this case, DTMF is sent in a channel that is not used by the actual voice traffic. Out-of-band signaling will be described in detail later in this chapter. Voice gateways should support supplementary services. Supplementary services are simply additional services that are available for VoIP. These services include hold capabilities, transfer capabilities, and conferencing capabilities, amongst many others. In Cisco VoIP networks, voice gateways should support Communications Manager redundancy. This allows the gateway to fail over to a secondary Communications Manager in the event that the primary Communications Manager fails.

11 NOTE: Cisco Unified Communications Manager is the new name for what was previously known and referred to as Cisco CallManager. These terms are interchangeable. Finally, voice gateways should support call survivability. This ensures that calls will not drop if the Communications Manager, to which either endpoint is registered, fails. Voice communication clients or endpoints include desktop IP phones, speaker phones, overhead paging systems, voice recording, operator console and wireless products. Voice Gateway Types Cisco voice gateway solutions are divided into three distinct categories. These categories are: 1. Cisco IOS Routers 2. Standalone Voice Gateways 3. Switch Modules Almost all Cisco IOS routers can be used as voice gateways. Cisco IOS routers have the capability to support both analog and digital ports. Common examples of Cisco IOS routers that support voice gateway functionality include Cisco Integrated Series routers, i.e. the Cisco 1800, 2800, and 3800 series routers, and the Cisco 2600XM Series Multiservice Router platforms. Standalone voice gateways can be used only as voice gateways. Whereas routers can be used for other purposes, such as routing and firewall capabilities, standalone voice gateways are dedicated voice gateways that support both analog and digital communications. Analog voice gateways connect to the PSTN or a PBX via analog ports. Additionally, analog voice gateways are also used to connect to various analog devices such as fax machines and modems. Examples of analog voice gateways include the Cisco Analog Telephone Adapter (ATA) series devices and the Cisco Voice Gateway (VG) 200 series voice gateways. Digital voice gateways, such as the Cisco AS5000 series digital voice gateways, connect VoIP networks to the PSTN or a PBX system via their digital trunk ports. Cisco also provides modules that support analog and digital ports which can be integrated into the Cisco Catalyst 6500 series switches and 7600 series routers. These modules include the 6600 series modules, which have now been replaced by corresponding Cisco Catalyst 6500 Series Communication Media Module (CMM) modules. The Cisco Communication Media Module for Catalyst 6500 Series and Cisco 7600 Series supports interconnectivity between the PSTN and traditional PBX systems and IP communication networks and Voice over IP networks.

12 DSP Resources Digital Signal Processor (DSP) resources are used in various Cisco voice gateways to convert analog voice signals to digital transmission across an IP network, and to convert back to analog once the packet has arrived at the destination router. DSP resources provide four critical functions in Cisco voice gateway solutions: Voice Termination Transcoding Conferencing Media Termination Point Voice termination is the process of digitizing and packetizing the voice audio stream on a TDM interface. A DSP is required to convert the traditional audio stream from the PSTN to a Voice over IP (VoIP) stream. DSPs have the capability to support multiple calls; however, the quantity of DSP resources that should be used in contingent on the complexity of the codec being used. In addition to voice termination, DSP resources also provide echo cancellation, Voice Activity Detection (VAD), and jitter management. Transcoders perform real-time translation or digitized voice from one codec to another. Because the various types of codecs, which will be described in detail later in this chapter, are not compatible with each other, transcoders are used to translate from one codec region to another. DSP resources used for Transcoding must be registered to the Cisco Unified Communications Manager via the SCCP protocol, which will be described in detail later in this guide. There are two types of conferencing available in Cisco VoIP solutions: software-based and hardware-based. Both hardware and software conference bridges can be active at the same time. Software and hardware conference devices differ in the number of streams and the types of codec they support. For software conference devices, you can adjust the number of streams. Hardware conference devices, however, support a fixed number of streams. Software-based conferencing is performed in software; however, hardware-based conferencing requires the use of DSPs registered to the Cisco Unified Communications Manager using the SCCP protocol. A Media Termination Point, or MTP, allows the Cisco CallManager to extend supplementary services, such as hold and transfer, to calls routed through an H.323 endpoint or an H.323 gateway. MTPs provide these supplementary services if H.323v2 is not supported end to end. As is the case with conferencing, there are software-based MTPs and hardware-based MTPs. Softwarebased MTPs are supported in Cisco voice gateways and do not require the use of DSP resources; however, hardware-based MTPs require the use of DSP resources. Again, these resources must be registered to the Cisco Unified Communications Manager via SCCP. DSP Types Cisco voice gateways support various types of DSP models, and each model has different capabilities. Some gateways and Network Modules (NM) use modular DSPs, which are referred to as Packet Voice DSP Modules (PVDM), while others use fixed DSPs, such as those found in NM-HD- 2V modules. The recommended DSP types to use are the C5510 DSPs. These high-density packet voice DSP modules are available in five versions, which are PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64. The following table shows the number of voice channels and codecs that each PVDM2 module supports: PVDM2 Name PVDM2-8 PVDM2 Description 8-channel packet fax and voice DSP Maximum Channels in G.711 Maximum Channels in High-Complexity Codecs Maximum Channels in Medium-Complexity Codecs

13 PVDM2-16 PVDM2-32 PVDM2-48 PVDM2-64 module 16-channel packet fax and voice DSP module 32-channel packet fax and voice DSP module 48-channel packet fax and voice DSP module 64-channel packet fax and voice DSP module The PVDM2 modules support the G.723.1, G.728, G.729, G.729b, ilbc, and Modem Relay highcomplexity codecs and the G.7112, g.729a, G.729ab, G.726, G.722, and Fax Relay mediumcomplexity codecs. PVDM2 can support a higher density (number) of G.711 calls than that of other medium-complexity codecs. Codecs will be described later in this chapter. NOTE: Calculating the number of DSP resources required is a challenging task that is beyond the scope of this course. Keep in mind that although the calculation can be performed manually, Cisco recommends using the online DSP calculator found directly on the Cisco website. The following table lists the different platforms on which the PVDM2 modules are supported: PVDM2 Name PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64 PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64 PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64 PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64 PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, and PVDM2-64 Platform Support NM-HDV2, NM-HDV2-1T1/E1, and NM-HDV2-2T1/E1 Cisco 2801, 2811, 2821, and 2851 Integrated Services Routers EVM-HD-8FXS/DID, EM-HDA-8FXS, and EM-4BRI- NT/TE Cisco 3825 and 3845 Integrated Services Routers Cisco 2901, 2911, 2921, 2951, 3925, 3945 Integrated Service Routers The show voice dsp command can be used to view the current status or selective statistics of digital signal processor (DSP) voice channels as illustrated in the following output: R1#show voice dsp FLEX VOICE CARD *DSP VOICE CHANNELS* DSP TX/RX TYPE COUN T DSP NU M C H CODEC DSPWARE VERSION CURR STATE BOOT STATE PAK RST AI VOICEPORT TS PACK ABRT C g711ulaw busy idle 0 0 1/0: C g711ulaw busy idle 0 0 1/0: *DSP SIGNALING CHANNELS* 31041/ / 26555

14 DSP TX/RX TYPE COUN T DSP NU M C H CODEC DSPWARE VERSION CURR STATE BOOT STATE PAK RST AI VOICEPORT TS PACK ABRT C {flex} alloc idle 0 0 0/0/ /0 C {flex} alloc idle 0 0 0/0/ /0 C {flex} alloc idle 0 0 0/0/ /0 C {flex} alloc idle 0 0 0/0/ /0 C {flex} alloc idle 0 0 0/1/ /0 C {flex} alloc idle 0 0 0/1/ /0 C {flex} alloc idle 0 0 0/1/ /0 C {flex} alloc idle 0 0 0/1/ /0 C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: / / / / / / / / / / / / / / / / / / / /65 71

15 C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 1/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: / / / / / / / / / / / / / / / / / / / / / / / / / / /60 24 C {flex} alloc idle 0 0 2/0: /59

16 C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: C {flex} alloc idle 0 0 2/0: END OF FLEX VOICE CARD / / / / / / / / / / FLEX VOICE CARD *DSP VOICE CHANNELS* DSP TX/RX TYPE COUN T DSP NU M C H CODEC *DSP SIGNALING CHANNELS* DSP TX/RX TYPE COUN T DSP NU M C H CODEC DSPWARE VERSION DSPWARE VERSION CURR STATE CURR STATE BOOT STATE BOOT STATE PAK RST AI VOICEPORT TS PACK ABRT PAK RST AI VOICEPORT TS PACK ABRT END OF FLEX VOICE CARD FLEX VOICE CARD *DSP VOICE CHANNELS* DSP TX/RX TYPE COUN T DSP NU M C H CODEC *DSP SIGNALING CHANNELS* DSP TX/RX TYPE COUN T DSP NU M C H CODEC DSPWARE VERSION DSPWARE VERSION CURR STATE CURR STATE BOOT STATE BOOT STATE PAK RST AI VOICEPORT TS PACK ABRT PAK RST AI VOICEPORT TS PACK ABRT

17 END OF FLEX VOICE CARD Voice DSPs can be used both for media and signaling channel resources. Media channels support the actual codec used for a live call while signaling channels are used by analog and CAS voiceports to monitor for and signal line events such as off-hook and on-hook. These concepts will be described in detail later in this chapter. In the output above, the *DSP VOICE CHANNELS* section provides information on DSP channels being used for media on the particular slot. The *DSP SIGNALING CHANNELS* provides information on DSP channels being used for signaling on the particular slot. The remainder of the pertinent columns are listed and described in the following table: Field DSP TYPE DSP NUM CH CODEC DSPWARE VERSION CURR STATE BOOT STATE RST AI VOICEPORT TS PAK ABRT TX/RX PAK COUNT Description This field specifies the DSP architecture type. This can read C542, C549, C5421, or C5510. This field specifies the unique identifier for the DSP in the pool of DSPs installed in the particular slot. This field uniquely identifies the channel on each DSP. Each DSP has logical channels capable of supporting media or signaling, up to 16 per C5510 DSP. Which codec is currently supported by the DSP media channel. This field specifies the DSPware version, which should always be consistent with the IOS version being used. This field specifies the current state of the DSP. This field specifies the boot state of the DSP. This field specifies the number of DSP ReSeTs counted. This field specifies the number of DSP Alarm Indicators counted. This field specifies the the voice port identifier associated with the DSP media or signaling channel. This field is used for digital T1/E1 CAS or PRI voice-ports, and BRI voice-ports, and specifies the TimeSlot involved. This field specifies the count of ABoRTed voice packets. This field specifies the count of transmitted and received voice packets. Cisco voice gateways determine how many voice calls each DSP can terminate based on codec complexity. For example, if the codec complexity is set to medium, the number of voice ports that can be terminated per DSP would be higher than if the codec complexity was set to high. Only voice cards that have C5510 DSPs support flex complexity, as illustrated in the output printed above. Flex complexity allows a variable number of calls per DSP based on voice gateway runtime calculations. Flex complexity is the default complexity for all voice cards that have C5510 DSPs. However, it should be noted that while flex complexity allows the DSPs to support the maximum number of voice calls, it also introduces the possibility of oversubscribing the DSP resources. It is important to understand the effect this may have in a production environment because if the DSPs are oversubscribed, calls will be connected, however, no audio path will exist when all available DSP channels are in use. DSP Sharing DSP sharing allows C5510 DSPs to terminate a voice call that is physically located in another hardware slot on modular platforms such as the Cisco 3800 ISR routers. DSP sharing reduces the possibility of oversubscription when using flex complexity and also makes it easier for the administrators to add DSPs to an existing voice gateway.

18 When DSP sharing is enabled, voice cards allow other voice cards to use their DSP resources. When a call comes in via a T1 or E1, the voice gateway must allocate a DSP for the call. If no local DSP is available for the call, the voice gateway searches for a remote DSP. After the gateway allocates a remote DSP, the DSP remains allocated for the duration of the call, even in the event that a local DSP becomes available. To enable DSP farm (sharing) services for a particular voice network module, use the dsp services dspfarm command in voice-port configuration mode. It is important to keep in mind that the DSPs that are installed on the main board of an ISR (e.g. Cisco 2800 and 3800 series) should always be configured under voice-card 0. Once configured, all other voice card that should participate in the DSP resource pool (including voice-card 0) should be configured with the dspfarm voice-card configuration command. The following output illustrates a sample configuration for basic DSP sharing on a Cisco 3800 ISR router with three PVDM2-64 modules: R1#show inventory include PVDM NAME: "PVDMII DSP SIMM with four DSPs", DESCR: "PVDMII DSP SIMM with four DSPs" PID: PVDM2-64, VID: V01, NAME: "PVDMII DSP SIMM with four DSPs", DESCR: "PVDMII DSP SIMM with four DSPs" PID: PVDM2-64, VID: V01, NAME: "PVDMII DSP SIMM with four DSPs", DESCR: "PVDMII DSP SIMM with four DSPs" PID: PVDM2-64, VID: V01, R1# R1# R1#conf t R1(config)#voice-card 0 R1(config-voicecard)#dsp services dspfarm R1(config-voicecard)#dspfarm R1(config-voicecard)#exit R1(config)#voice-card 1 R1(config-voicecard)#dspfarm R1(config-voicecard)#exit R1(config)#voice-card 2 R1(config-voicecard)#dspfarm R1(config-voicecard)#exit

19 R1(config)#exit This configuration can then be validated via the show running-configuration command: R1#show running-config begin voice-card 0 voice-card 0 dspfarm dsp services dspfarm! voice-card 1 dspfarm! voice-card 2 dspfarm Configuring Conferencing, MTP and Transcoding Resources Once DSP sharing has been enabled, the voice gateway can be configured to register these resources with Cisco Unified Communications Manager. While the configuration of configuration of these elements is beyond the scope of the CVOICE, the following gateway configuration illustrates the configuration of two hardware-based profiles: one for transcoding and the other for conferencing; and one software-based MTP profile.! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 10 associate application SCCP!

20 dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 9 associate application SCCP! In the output above, the conference profile has been defined with numerous codecs and allows up to a maximum of 10 conference sessions. The transcode profile is also defined with numerous codecs and allows up to 9 transcoding sessions. The associate application SCCP command instructs the voice gateway to register the resources with the Cisco Unified Communications Manager via SCCP. The following voice gateway configuration output illustrates a basic SCCP configuration to register the profiles illustrated above: interface GigabitEthernet0/0 description 'Voice Gateway interface for CUCM SCCP Registration" ip address ! sccp local GigabitEthernet0/0 sccp ccm identifier 1 version 7.0+ sccp! sccp ccm group 1 bind interface GigabitEthernet0/0 associate ccm 1 priority 1 associate profile 1 register CVOICE_CNF_001 associate profile 2 register CVOICE_TRN_001 The DSP show dspfarm dsp all command shows the DSP resources reserved for conferencing and transcoding based on the configuration implemented on the voice gateway: R1#show dspfarm dsp all SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED UP N/A FREE xcode

21 UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE xcode UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf UP N/A FREE conf Total number of DSPFARM DSP channel(s) 19 The show dspfarm profile command shows the current status of the DSP sharing profile configuration and whether or not they are associated with the Cisco Unified Communications Manager which they have been configured to register to: R1#show dspfarm profile Dspfarm Profile Configuration Profile ID 2, Service TRANSCODING, Resource ID 1 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 9 Number of Resource Available : 9 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30

22 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 Dspfarm Profile Configuration Profile ID 1, Service CONFERENCING, Resource ID 2 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 10 Number of Resource Available : 10 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30, Transcoder: Not Required Codec : g711alaw, Maximum Packetization Period : 30, Transcoder: Not Required Codec : g729ar8, Maximum Packetization Period : 60, Transcoder: Not Required Codec : g729abr8, Maximum Packetization Period : 60, Transcoder: Not Required Codec : g729r8, Maximum Packetization Period : 60, Transcoder: Not Required Codec : g729br8, Maximum Packetization Period : 60, Transcoder: Not Required Dspfarm Profile Configuration Finally, the show sccp ccm group command can be used to view the registration of the configured profiles to the Cisco Unified Communications Manager as follows: R1#show sccp ccm group CCM Group Identifier: 1 Description: None Binded Interface: GigabitEthernet0/0, IP Address: Associated CCM Id: 1, Priority in this CCM Group: 1 Associated Profile: 1, Registration Name: CVOICE_CNF_001

23 Associated Profile: 2, Registration Name: CVOICE_TRN_001 Registration Retries: 3, Registration Timeout: 10 sec Keepalive Retries: 3, Keepalive Timeout: 30 sec CCM Connect Retries: 3, CCM Connect Interval: 10 sec Switchover Method: GRACEFUL, Switchback Method: GRACEFUL_GUARD Switchback Interval: 10 sec, Switchback Timeout: 7200 sec Signaling DSCP value: cs3, Audio DSCP value: ef When configuring DSP farming, it is important to remember that you can configure either hardware-based or software-based MTP profiles on the router; however, this is not applicable to transcoding or conferencing profiles, which are only hardware-based and can never be performed in software. The options available for MTP profiles are illustrated below: R1(config)#dspfarm profile 3 mtp R1(config-dspfarm-profile)#maximum sessions? hardware Hardware MTP software Software MTP The following configuration illustrates the configuration of a software-based MTP profile, which is then registered to the Cisco Unified Communications Manager: interface GigabitEthernet0/0 description 'Voice Gateway interface for CUCM SCCP Registration" ip address ! sccp ccm group 1 bind interface GigabitEthernet0/0 associate ccm 1 priority 1 associate profile 3 register CVOICE_001! dspfarm profile 3 mtp codec g711ulaw maximum sessions software 200 associate application SCCP

24 This configuration can then be validated via the show dspfarm profile [number] command: R1#show dspfarm profile 3 Dspfarm Profile Configuration Profile ID 3, Service MTP, Resource ID 3 Profile Description : Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : NONE Status : NONE Number of Resource Configured : 200 Number of Resource Available : 200 Hardware Configured Resources : 0 Hardware Available Resources : 0 Software Resources : 200 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 In Cisco Unified Communications Manager, this association can be validated under Media Resources > Media Termination Point, as illustrated below on a CUCM 7.0 server:

25 NOTE: The screenshot above is illustrated only as an example. Also remember that you are not required to perform any CUCM or DSP configuration in the CVOICE 6.0 certification exam. This information has been provided above to reinforce the concepts described in this section.