Addressing by E164 telephone number

MGCP defines an environment for controlling telephony gateways from a centralized call control component known as a call agent. An MGCP gateway handles the translation of audio between the telephone SCN and the packet-switched network of the Internet. Gateways interact with a call agent that performs signaling and call processing. IETF RFC 2705 defines MGCP. RFC 2805 defines an architecture for MGCP. These IETF standards describe MGCP as a centralized device control protocol with simple...

Admission Control

All rights reserved. Admission control has at least two aspects authorization and bandwidth management. Access to a network should not imply permission to use the resources of the network. Common control limits access to the resources by checking the intentions and credentials of users before authorizing them to proceed. Bandwidth is a finite resource. Appropriate bandwidth management is essential to maintaining voice quality. Allowing too many voice calls over an IP...

All ephonedns must be the same type singleline or dualline

The partially automated setup is exactly like a manual setup, without having to configure ephones. The ephones can be detected automatically and assigned an ephone-dn from a range of configured ephone-dns. This allows for the deployment of many phones without the work of configuring every phone manually. This automatic assignment is done through the use of the auto assign command. To automatically assign ephone-dn tags to Cisco IP phones as they register for service with the Cisco CME router,...

Allows a file in flash to be downloadable with TFTP

The command tftp-server flashfilename allows the file specified that resides in flash to be downloaded via TFTP. In Cisco CME the firmware files need to be configured to be available through TFTP. The example above shares firmware for the 7910, 7940-7960, and the 7920 IP phones. The command telephony-service enters the telephony service mode where much of the configuration of the Cisco CME system is entered. Two of the first commands that will want to be entered are the max-dn and max-ephone....

Analog vs Digital

A gateway is a device that translates one type of signal to a different type of signal. There are different types of gateways, including the voice gateway. A voice gateway is a router or switch that converts IP voice packets to analog or digital signals that are understood by TDM trunks or stations. Gateways are used in several situations for example, to connect the PSTN, a PBX, or a key system to a VoIP network.

AnnexB variant may be applied to either

G.729, G.729 Annex-A (G.729A), G.729 Annex-B (G.729B), and G.729A Annex-B (G.729AB) are variations of CS-ACELP. There is little difference between the ITU recommendations for G.729 and G.729A. All of the platforms that support G.729 also support G.729A. G.729 is the compression algorithm that Cisco uses for high-quality 8-kbps voice. When properly implemented, G.729 sounds as good as the 32-kbps ADPCM. G.729 is a high-complexity, processor-intensive, compression algorithm that monopolizes...

Applying QoS for Endto End Improvement of Voice Quality

Voice features for Cisco IOS QoS are deployed at different points in the network and designed for use with other QoS features to achieve specific goals, such as control over jitter and delay. This topic lists the network areas in which Cisco IOS QoS is implemented. Cisco IOS software includes a complete set of features for delivering QoS throughout the network. Following are Cisco IOS features that address the voice packet delivery requirements of end-to-end QoS and service differentiation In...

Assigns a primary extension number to an ephonedn

When an ephone-dn is configured with a single line, one virtual voice port is configured and, since only a single line exists, only one call to or from the ephone-dn can be active. The second call that arrives while a call is active will receive whatever is the defined busy treatment. Configuring an ephone-dn in this fashion mimics typical functionality of a keyswitch line. This ephone-dn will lack some of the advanced PBX features.

Associates a name with an extension ephonedn

The directory command is used to set the system-wide setting for how names will be displayed in the CME directory. The default is first name first. Entries that represent non-IP phones controlled by Cisco CME are entered into the directory from the CLI using the directory entry command. This can also be done by using the GUI, as seen in the previous page. The name command is how an identity is associated to the ephone-dn in Cisco CME. Enter the name in the same order that was defined using the...

ATM trunk side interface

Endpoints represent the point of interconnection between the packet network and the traditional telephone network. Endpoints can be physical, representing an FXS port or a channel in a T1 or E1, or they can be logical, representing an attachment point to an announcement server. To manage an endpoint, the call agent must recognize the characteristics of an endpoint. To aid in this process, endpoints are categorized into several types. The intent is to configure a call agent to manage a type of...

Automated Setup Cont Results

Flash is searched and if firmware is found it will be Creates SEP XML files at boot up and load to RAM Firmware is searched and if MoH is found this entry is made The selected number of ephone-dns are configured This shows the results of the automated setup. Note ITS is the initial name of what is now called Cisco CME, and still appears in some of the configuration that is created with the automated setup. network 10.90.0.0 255.255.255.0 default-router 10.90.0.1 option 150 ip 10.90.0.1...

Automation with Cisco AutoQoS Diff Serv Functions Automated

Link Fragmentation & Interleaving Classification of VoIP based on packet attributes or port trust Set L3 L2 attributes to categorize packets into a class Provide EF treatment to voice & BE treatment to data Shape to CIR to prevent burst & smooth Traffic to Configured Rat Reduce the VoIP bandwidth requirement Reduce jitter experienced by voice packets 2005 Cisco Systems, Inc. All rights re Cisco AutoQoS performs the following functions Automatically classify RTP payload and VoIP control...

AutoQoS

This topic describes the basic purpose and function of AutoQoS. One command per interface to enable and configure QoS 2005 Cisco Systems, Inc. All rights re 2005 Cisco Systems, Inc. All rights re AutoQoS enables customer networks the ability to deploy QoS features for converged IP telephony (IPT) and data networks much faster and more efficiently. It simplifies and automates the Modular QoS CLI (MQC) definition of traffic classes, and the creation and configuration of traffic policies (Cisco...

AutoQoS Router Platforms

Cisco 1760, 2600, 3600, 3700 and 7200 Series Routers User can meet the voice QoS requirements without extensive knowledge about Underlying technologies (ie PPP, FR, ATM) AutoQoS lends itself to tuning of all generated parameters & configurations 2005 Cisco Systems, Inc. All rights re Initial support for AutoQoS includes the Cisco 2600, 2600-XM, 3600, 3700, and 7200 series routers. Support for additional platforms will become available. Cisco AutoQoS VoIP feature is supported only on the...

Auxiliary VLANs Cont IP Addressing Deployment Options

IP Phone + PC on separate switch ports 171.68.249.101 I 171.68.249.100 IP Phone + PC on separate switch ports 10.1.1.1 171.68.249.100 IP Phone + PC on separate switch ports 10.1.1.1 171.68.249.100 Cisco IP Phones require network IP addresses. Cisco makes the following recommendations for IP addressing deployment Continue to use existing addressing for data devices (PCs, workstations, and so forth). Add IP Phones with Dynamic Host Configuration Protocol (DHCP) as the mechanism for obtaining...

Availability and fault tolerance

When designing a large-scale numbering plan, you must adhere to the following attributes Logic distribution Good dial plan architecture relies on the effective distribution of the dial plan logic among the various components. Devices that are isolated to a specific portion of the dial plan reduce the complexity of the configuration. Each component focuses on a specific task accomplishment. Generally, the local switch or gateway handles details that are specific to the local point of presence...

Average Jitter Statistics

show call active voice < output omitted> ConnectionId 0xECDE2E7B 0xF46A003F 0x0 0x47070A4 IncomingConnectionId 0xECDE2E7B 0xF46A003F 0x0 0x47070A4 SessionProtocol cisco SessionTarget OnTimeRvPlayout 482350 GapFillWithSilence 1040 ms GapFillWithPrediction 3160 ms GapFillWithInterpolation 0 ms GapFillWithRedundancy 0 ms HiWaterPlayoutDelay 70 ms LoWaterPlayoutDelay 29 ms ReceiveDelay 43 ms LostPackets 0 EarlyPackets 0 LatePackets 105 The sample output in this figure displays average jitter...

Bandwidth Requirements in VoIP

This topic describes the bandwidth that each coder-decoder (codec) uses and illustrates its impact on total bandwidth. One of the most important factors for the network administrator to consider while building voice networks is proper capacity planning. Network administrators must understand how much bandwidth is used for each Voice over IP (VoIP) call. With a thorough understanding of VoIP bandwidth, the network administrator can apply capacity-planning tools. Following is a list of codecs and...

Bandwidth Requirements in VoIP Data Link Overhead

This topic lists overhead sizes for various Layer 2 protocols. This topic lists overhead sizes for various Layer 2 protocols. Another contributing factor to bandwidth is the Layer 2 protocol used to transport VoIP. VoIP alone carries a 40-byte IP User Datagram Protocol Real-Time Transport Protocol (IP UDP RTP) header, assuming uncompressed RTP. Depending on the Layer 2 protocol used, the overhead could grow substantially. The larger the Layer 2 overhead, the more bandwidth required to transport...

Bandwidth Requirements in VoIP Effect of VAD

This topic describes the effect of voice activity detection (VAD) on total bandwidth. On average, an aggregate of 24 calls or more may contain 35 percent silence. With traditional telephony voice networks, all voice calls use 64-kbps fixed-bandwidth links regardless of how much of the conversation is speech and how much is silence. With Cisco VoIP networks, all conversation and silence is packetized. VAD suppresses packets of silence. Instead of sending VoIP packets of silence, VoIP gateways...

Bandwidth Requirements in VoIP Impact of Voice Samples

This topic illustrates the effect of voice sample size on bandwidth. This topic illustrates the effect of voice sample size on bandwidth. Voice sample size is a variable that can affect total bandwidth used. A voice sample is defined as the digital output from a codec digital signal processor (DSP) that is encapsulated into a protocol data unit (PDU). Cisco uses DSPs that output samples based on digitization of 10 ms worth of audio. Cisco voice equipment encapsulates 20 ms of audio in each PDU...

Based on other well known protocols

SIP was designed as a multimedia protocol that could take advantage of the architecture and messages found in popular Internet applications. By using a distributed architecture with URLs for naming and text-based messaging SIP attempts to take advantage of the Internet model for building VoIP networks and applications. In addition to VoIP, SIP is used for videoconferencing and instant messaging. As a protocol, SIP only defines how sessions are to be set up and torn down. It utilizes other IETF...

Baselining Input and Output Power Levels

Considerations for baselining input and output power levels are as follows Analog voice routers operate best when the receive level from an analog source is set at approximately -3 dB. In the United States and most of Europe, the receive (transmit) level that is normally expected for an analog telephone is approximately -9 dB. In Asian and South American countries, receive levels are closer to -14 dB. To accommodate these differences, the output levels to the router are set over a wide range....

Basic Call Setup

The figure shows the three major steps in an end-to-end call. These steps include 1. Local signaling originating side The user signals the switch by going off hook and sending dialed digits through the local loop. 2. Network signaling The switch makes a routing decision and signals the next, or terminating, switch through the use of setup messages sent across a trunk. 3. Local signaling terminating side The terminating switch signals the call recipient by sending ringing voltage through the...

Basic MGCP Concepts

This topic introduces the basic MGCP concepts. This topic introduces the basic MGCP concepts. The basic MGCP concepts are listed below Calls and connections Allow end-to-end calls to be established by connecting two or more endpoints Events and signals Fundamental MGCP concept that allows a call agent to provide instructions for the gateway Packages and digit maps Fundamental MGCP concept that allows a gateway to determine the call destination

Basic Voice Encoding Converting Digital to Analog

This topic describes the process of converting digital signals back to analog signals. After the receiving terminal at the far end receives the digital PCM signal, it must convert the PCM signal back into an analog signal. The process of converting digital signals back into analog signals includes the following Decoding The received eight-bit word is decoded to recover the number that defines the amplitude of that sample. This information is used to rebuild a PAM signal of the original...

Best practice is to configure a DHCP scope for the IP phones

DHCP is a very common and familiar protocol to many network administrators. A scope will be defined per subnet and is used to hand out IP addresses from a pool of available addresses, along with a subnet mask. Optionally, other values like the default gateway and DNS can be assigned to the scope by setting option values if desired. For example, the default gateway option is 003 and DNS is 006. These option values can include values specific to an implementation and can be customized by the...

BRI Reference Points

Given all the ISDN interface abbreviations such as T, S, U, S T, and so on, what do all of these components and reference points look like in practice When creating a network, connect the network termination 1 (NT1) to the wall jack with a standard two-wire connector, then to the ISDN phone, terminal adapter, Cisco ISDN router, and maybe a fax with a four-wire connector. The S T interface is implemented using an eight-wire connector (two pairs for data transmission and two pairs for providing...

Builds the specific XML files necessary for the IP phones

All rights re To build the XML configuration files that are required for IP phones used with Cisco CME 3.1, or later versions, use the create cnf-files command in telephony-service configuration mode. When this command is entered the file XMLDefault.cnf.xml is generated with that appropriate settings including the firmware defined by the load command, the IP address for new IP phones to register to and the TCP port those messages will arrive on.

Cac

WAN bandwidth can support only n calls. What happens when n + 1 calls are attempted X4111 quality for all calls. x3111 X4111 quality for all calls. x3111 CAC is required to ensure that network resources are not oversubscribed. CAC could be described as a way to protect voice from voice. Calls that exceed the specified bandwidth are either rerouted using an alternative route such as the PSTN, or a fast busy tone is returned to the calling party. This way the next voice call does not degrade the...

Call Agents

Residential Gateway Residential Gateway Residential Gateway Residential Gateway A call agent, or Media Gateway Controller (MGC), represents the central controller in an MGCP environment. A call agent exercises control over the operation of a gateway and its associated endpoints by requesting that a gateway observe and report events. In response to the events, the call agent instructs the endpoint what signal, if any, the endpoint should send to the attached telephone equipment. This requires a...

Call Control

0 FXS Network S * Although different protocols address call control in different ways, they all provide a common set of services. The following are basic components of call control Call setup Checks call-routing configuration to determine the destination of a call. The configuration specifies the bandwidth requirements for the call. When the bandwidth requirements are known, Call Admission Control (CAC) determines if sufficient bandwidth is available to support the call. If bandwidth is...

Call Control Approach to CAC

All rights re CAC, as part of call control services, functions on the outgoing gateway. CAC bases its decision on nodal information, such as the state of the outgoing LAN or WAN link. If the local packet network link is down, there is no point in executing complex decision logic based on the state of the rest of the network, because that network is unreachable. Local mechanisms include configuration items that disallow all calls that exceed a specified number.

Call Control Models

This topic describes several call control models and their corresponding protocols. This topic describes several call control models and their corresponding protocols. The following call control models and their corresponding protocols exist or are in H.323 International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Recommendation H.323 describes the architecture to support multimedia communications over networks without quality of service (QoS) guarantees. Originally...

Call Flow with a Gatekeeper

All rights re 2005 Cisco Systems, Inc. All rights re The exchanges in the figure illustrate the use of a gatekeeper by both endpoints. In this example, both endpoints have registered with the same gatekeeper. Call flow with a gatekeeper proceeds as follows 1. The gateway sends an ARQ to the gatekeeper to initiate the procedure. The gateway is configured with the domain or address of the gatekeeper. 2. The gatekeeper responds to the admission request with an ACF. In the...

Call Flow with Multiple Gatekeepers

All rights reserved. 2005 Cisco Systems, Inc. All rights reserved. The figure illustrates a call setup involving two gatekeepers. In this example, each endpoint is registered with a different gatekeeper. Notice the changes in the following call setup procedure 1. The originating endpoint sends an admission request to its gatekeeper requesting permission to proceed and asking for the session parameters for the terminating endpoint. 2. The gatekeeper for the originating...

Call Flows

The figure illustrates a dialog between a call agent and two gateways. Although the gateways in this example are both residential gateways, the following principles of operation are the same 1. The call agent sends a notification request (RQNT) to each gateway. Because they are residential gateways, the request instructs the gateways to wait for an off-hook transition (event). When the off-hook transition event occurs, the call agent instructs the gateways to supply dial tone (signal). The call...

Call Forwarding Cont Forwarding a Call from an IP phone

Forward all, busy, and no answer all in the phone user Web pages There are five call forwarding commands that can be configured from the command line of the Cisco CME router. These commands are the following Call forward all (CLI, GUI, Phone) Call forward busy (CLI, GUI) Call forward no answer (CLI, GUI) Call forward max-length (CLI)

Call Setup

A fundamental objective of VoIP call control is to initiate communication between VoIP endpoints. This topic discusses the role of call control in establishing RTP sessions and negotiating features during the call setup procedure. An audio path of a VoIP call leg is dependent on the creation of RTP sessions. These RTP sessions transport voice unidirectionally, so that bidirectional voice uses two RTP sessions. (In principle, if voice is needed in one direction only, as in the case of a recorded...

Call Status

All rights reserved. Several of the responsibilities that are assigned to call administration and accounting are dependent on access to current call status information or records of changes in the call status. Call status has both historical and instantaneous (real-time) benefits. Call detail records have consequential benefits in terms of distributing costs and planning capacity. Call status provides an instantaneous view of the calls that are in progress. This view...

Call Transfer

This topic describes the Cisco CME transfer commands. Transferring a caller to another directory number is a very common occurrence in the enterprise. The person on the IP phone can initiate a transfer by using the functions displayed on the IP phone display. To transfer a caller the user can initiate the transfer by pressing the Trnsfer softkey button and dialing the number that the call will be transferred to. Depending on the configuration deployed on the Cisco CME system the call will...

Calling and Directory Features

This topic describes calling and directory features. Directory can be accessed by pressing the directory button When a user does not know the number of another subscriber of commonly used external number the corporate directory that resides in the Cisco CME can be accessed and the number looked up and connected. The directory of the Cisco CME is built and stored on the router from the configuration. It can be accessed by the phone users by pressing the directory button (assuming the url command...

Calls and Connections

All rights reserved. 2005 Cisco Systems, Inc. All rights reserved. End-to-end calls are established by connecting two or more endpoints. To establish a call, the call agent instructs the gateway that is associated with each endpoint to make a connection with a specific endpoint or an endpoint of a particular type. The gateway returns the session parameters of its connection to the call agent, which in turn sends these session parameters to the other gateway. With this...

CDP must be enabled for AutoQoS to function properly

To configure the QoS settings and the trusted boundary feature on the Cisco IP Phone, you must enable Cisco Discovery Protocol (CDP) version 2 or later on the port. If you enable the trusted boundary feature, a syslog warning message displays if CDP is not enabled or if CDP is running version 1. You need to enable CDP only for the ciscoipphone QoS configuration CDP does not affect the other components of the automatic QoS features. When you use the ciscoipphone keyword with the port-specific...

Central Office Switches

The figure shows a typical CO switch environment. The CO switch terminates the local loop and makes the initial call-routing decision. The call-routing function forwards the call to one of the following Another end-user telephone, if it is connected to the same CO The CO switch makes the telephone work with the following components Battery The battery is the source of power to both the circuit and the telephone. It determines the status of the circuit. When the handset is lifted to let current...

Centralized Call Control

CA1 signals R1 to send dial tone. 4. CA1 sends setup message to CA2, 1. R1 alerts CA1 of off hook state, 2. CA1 signals R1 to send dial tone. 4. CA1 sends setup message to CA2, 2. CA2 determines call destination is R2, 3. CA2 signals R2 to send ring signal out specific port. Centralized call control allows an external device (call agent) to handle the signaling and call processing, leaving the gateway to translate audio signals into voice packets after call...

Channel Associated Signaling Systems

Because the signaling occurs within each DS0, it is referred to as in band. Also, because the use of these bits is exclusively reserved for signaling each respective voice channel, it is referred to as CAS. SF has a 12-frame structure and provides AB bits for signaling. ESF has a 24-frame structure and provides ABCD bits for signaling. Tones, such as dual tone multifrequency (DTMF) addressing or call progress, can be carried in the audio path. However, other CAS signals must be carried via the...

Channel Associated Signaling Systems E1

In E1 framing and signaling, 30 of the 32 available channels, or time slots, are used for voice or data. Framing information uses time slot 1 (channel 0), while time slot 17 (channel 16) is used for signaling by all the other time slots. This signaling format is also known as CAS because each bearer channel has specific bits in the 17th timeslot assigned for signaling. However, this implementation of CAS is considered out of band because the signaling bits are not carried within the voice...

Circuit IDchannel t1toSJ17mgcpgatewayciscocom

When interacting with a gateway, the call agent directs its commands to the gateway for the express purpose of managing an endpoint or a group of endpoints. An endpoint identifier, as its name suggests, identifies endpoints. Endpoint identifiers consist of two parts a local name of the endpoint in the context of the gateway and the domain name of the gateway itself. The two parts are separated by an at sign ( ). If the local part represents a hierarchy, the subparts of the hierarchy are...

Cisco Call Manager Express Restrictions Cont

Operation of multiple independent clients (e.g. one client per phone line) Windows phone dialer Outlook contact dialer Third party applications Multiple-user or multiple-call handling (Required for ACD) Cisco CME does not support TAPI v2.1. Cisco CME TAPI implements only a small subset of TAPI functionality. It does support operation of multiple independent clients (for example, one client per phone line) but not full support for multiple-user or multiple-call handling, which is required for...

Cisco CME can register to a H323 gatekeeper thereby ensuring the WAN is not oversubscribed

Register Extension number and or E.164 number Register Extension number and or E.164 number 2005 Cisco Systems, Inc. All rights reserved. The Cisco CME system can be configured to register the ephone-dns with a H.323 Gatekeeper. In addition, the IP phone may have both an extension number and an E. 164 number defined, and one or both of the numbers may be registered with the H.323 Gatekeeper. H.323 can also be used to allow one Cisco CME to communicate with another Cisco CME or Voice Gateways. A...

Cisco CME does not support remotely registered phones

Cisco CME does not support remotely registered phones via a WAN or virtual private network (VPN) connection because the Skinny interface does not have the necessary set of QoS tools these tools have been built into the H.323 VoIP interface to cope with operating across nonlocal networks. Cisco CME also does not support bandwidth control or accounting, RSVP, or the max-conn attribute for remotely registered SCCP phones via a WAN or virtual private network (VPN) connection. Each remote site...

Cisco CME provides three levels of HTTP based GUI access

The Cisco CME GUI provides a Web-based interface to manage most system-wide and phone-based features. In particular, the GUI facilitates the routine adds and changes associated with employee turnover, allowing these changes to be performed by non-technical staff. The GUI provides three levels of access to support the following user classes System Administrator - Able to configure all system wide and phone-based features. This person is familiar with Cisco IOS software and VoIP network...

Cisco Implementation of H323

All rights re Cisco provides support for all H.323 components. These H.323 components include the following H.323 terminals Cisco provides support for H.323 terminals in Cisco IP Phone. Gateways Cisco implements H.323 gateway support in Cisco voice-enabled routers (first available in Cisco IOS Release 11.3) Cisco SC2200 Signaling Controllers Cisco PGW 2200 PSTN gateways Voice-enabled Cisco AS5xx0 access servers Gatekeepers Cisco implements gatekeeper support in Cisco...

Class of Restriction

COR List on Incoming dial-peer or ephone-dn COR List on Outgoing dial-peer or ephone-dn The no (null) incoming COR condition has the highest COR priority The incoming COR list is a superset of the no (null) outgoing COR list Incoming COR applied is a superset of outgoing COR The incoming COR list is a superset of the outgoing COR list Incoming COR applied not a superset of outgoing COR The incoming COR list is NOT a superset of the outgoing COR list By default, an incoming call leg has the...

Class of Restriction Case Study

Class of Restriction Case Study - XYZ company The XYZ company wishes to prevent toll fraud by restricting the destinations on the PSTN that IP phones and analog phones attached to FXS port can call. There should be no restrictions internally everyone internal should be able to call anyone else internal All phones MUST be able to call 911 Within the XYZ company there are Lobby phones, Employee phones, Sales, and Executive phones The Lobby phone should be able to call only 911 on the PSTN The...

Class of Restriction COR

All rights re The executive can call the employee but the employee cannot call the executive The incoming COR Employee is not a superset of the Executive, so the call will not succeed The incoming COR Employee is not a superset of the Executive, so the call will not succeed 2005 Cisco Systems, Inc. All rights re

CO Switching Systems

Switching systems provide three primary functions Call setup, routing, and teardown Customer ID and telephone numbers CO switches switch calls between locally terminated telephones. If a call recipient is not locally connected, the CO switch decides where to send the call based on its call-routing table. The call then travels over a trunk to another CO or to an intermediate switch that may belong to an interexchange carrier (IXC). Although intermediate switches do not provide dial tone, they...

Coder Delay

The compression time for a Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP) process ranges from 2.5 to 10 ms, depending on the loading of the DSP. If the DSP is fully loaded with four voice channels, the coder delay will be 10 ms. If the DSP is loaded with one voice channel only, the coder delay will be 2.5 ms. For design purposes, use the worst-case time of 10 ms. Decompression time is roughly 10 percent of the compression time for each block. However, because there may...

Commands to Verify Voice Ports

Shows all voice port configurations in detail Shows one voice port configuration in detail Shows all voice port configurations in brief Shows all ports configured as busyout Shows the operational status of the controller There are six show commands for verifying the voice port and dial-peer configuration. These commands and their functions are shown in the figure.

Common Channel Signaling

Whereas CAS uses bit time slots assigned to each specific channel, CCS uses a common channel and protocol to setup calls for all the bearer channels. Using ISDN over E1 as an example, the signaling protocol Q931 would use timeslot 17 to exchange call-setup messages for any of the 30 bearer (B) channels. Examples of CCS signaling are as follows Proprietary implementations Some PBX vendors choose to implement a proprietary CCS protocol between their PBXs for T1 and E1. In this implementation,...

Compress the samples to reduce bandwidth multiplexing optional step

Digitizing speech was a project first undertaken by the Bell System in the 1950s. The original purpose of digitizing speech was to deploy more voice circuits with a smaller number of wires. This evolved into the T1 and E1 transmission methods of today. To convert an analog signal to a digital signal, you must perform these steps Note The last step is optional. The sampling rate must be two times the highest frequency to produce playback that appears neither choppy nor too smooth. Quantization...

Compress the samples to reduce bandwidth optional step

Digitizing speech was a project first undertaken by the Bell System in the 1950s. The original purpose of digitizing speech was to deploy more voice circuits with a smaller number of wires. This evolved into the T1 and E1 transmission methods of today. To convert an analog signal to a digital signal, you must perform these steps The sampling rate must be twice the highest frequency to produce playback that appears neither choppy nor too smooth. Quantization consists of a scale made up of eight...

Compression Bandwidth Requirements

The following three common voice compression techniques are standardized by the ITU-T PCM Amplitude of voice signal is sampled and quantized at 8000 times per second. Each sample is then represented by one octet (8 bits) and transmitted. For sampling, you must use either a-law or -law to reduce the signal-to-noise ratio. ADPCM The difference between the current sample and its predicted value (based on past samples). ADPCM is represented by 2, 3, 4, or 5 bits. This method reduces the bandwidth...

Configuration Parameters

FXS port configuration allows you to set parameters based on the requirements of the connection if default settings need to be altered or the parameters need to be set for fine-tuning. You can set the following configuration parameters signal Sets the signaling type for the FXS port. In most cases, the default signaling of loop start works well. If the connected device is a PBX or a key system, the preferred signaling is ground start. Modern PBXs and key systems do not normally use FXS ports as...

Configure the interface or controller settings

To configure an ISDN BRI interface on a router, global and interface configuration commands must be specified. Global configuration tasks include Select the switch type that matches the ISDN provider switch at the central office (CO). Set destination details. Indicate static routes from the router to other ISDN destinations. Specify the traffic criteria that initiate an ISDN call to the appropriate destination. Interface configuration tasks include Select the ISDN BRI port and configure an IP...

Configuring Administrative User Classes Cont

Select the phone of the user, then set credentials on the phone Select the phone of the user, then set credentials on the phone To set phone user credentials from the phone user Web pages, go to the Configure dropdown menu and select Phones. Either add a new phone or change an existing phone by selecting it. Scroll to the bottom of the page and in the Login Account area, define the user and password. Select the Change button to commit the changes. Configuring Administrative User Classes (Cont.)...

Configuring an MGCP Residential Gateway

Dial-peer voice 1 pots application MGCPAPP port 1 0 0 dial-peer voice 2 pots application MGCPAPP port 1 0 1 The figure highlights the commands required to configure an MGCP residential gateway. MGCP is invoked with the mgcp command. If the call agent expects the gateway to use the default port (UDP 2427), the mgcp command is used without any parameters. If the call agent requires a different port, then the port must be configured as a parameter in the mgcp command for example, mgcp 5036 would...

Configuring an MGCP Trunk Gateway

Ccm-manager-mgcp mgcp 4000 mgcp call-agent 209.165 The ccm-manager-mgcp command is required only if the call agent is a Cisco CallManager. The second example illustrates the configuration of a trunk gateway. Configuring trunk gateways requires the address or the name of the call agent, which is a requirement common to a residential gateway (RGW). The trunk package is the default for a trunk gateway and does not need to be configured. Again, other parameters are optional.

Configuring AutoQoS

All rights re 2005 Cisco Systems, Inc. All rights re Cisco AutoQoS is innovative technology that minimizes the complexity, time, and operating cost of QoS deployment. Cisco AutoQoS incorporates value-added intelligence into Cisco IOS software and Cisco Catalyst Operating Service software to provision and manage large-scale QoS deployments. The first phase of Cisco AutoQoS targets VoIP deployments for customers who want to deploy IP telephony, but who lack the expertise...

Configuring AutoQoS Prerequisites for Using AutoQoS

Cisco Express Forwarding (CEF) must be enabled at the interface or ATM PVC This feature cannot be configured if a QoS policy (service policy) is attached to the interface An interface is classified as low-speed if its bandwidth is less than or equal to 768 kbps. It is classified as high-speed if its bandwidth is greater than 768 kbps The correct bandwidth should be configured on all interfaces or sub-interfaces using the bandwidth command If the interface or sub-interface has a link speed of...

Configuring Auxiliary VLANs

All rights reserved. 2005 Cisco Systems, Inc. All rights reserved. All data devices typically reside on data VLANs in the traditional switched scenario. You may need a separate voice VLAN when you combine the voice network into the data network. The Catalyst software command-line interface (CLI) refers to this new voice VLAN as the auxiliary VLAN for configuration purposes. You can use the new auxiliary VLAN to represent other types of devices. Currently, the device is...

Configuring Auxiliary VLANs Router Configuration

Routing between the different VLANs requires a layer 3 router. The router will need to have an interface local to all of the VLANs to which it will route. The most efficient way to get multiple VLANs to the router is by connecting a trunk between the switch and the router. This configuration is known as router on a stick. The router will have one sub-interface local to each VLAN and only one VLAN can be assigned to that sub-interface.

Configuring H323 Gatekeepers

This topic illustrates the gatekeeper configuration for a two-zone, two-gatekeeper scenario. The gatekeeper application is enabled with the gatekeeper command. For this example, the gateways are configured to withhold their E.164 addresses, so the gatekeepers must define the addresses locally. This is done with the zone prefix command. In the example, each gatekeeper has two zone prefix commands, the first pointing to the other gatekeeper and the second pointing to the local zone (meaning the...

Configuring the Gateways

Interface Ethernet0 0 ip address 10.52.218.49 255.255,255,0 h323-gateway voip interface h323-gateway voip id gk-zonel.test.com ipaddr 10.52.218.47 1718 h323-gateway voip h323-id gw_l h323-gateway voip bind srcaddr 10.52.218.49 1 dial-peer voice 1 voip destination-pattern 16 session target ras dial-peer voice 2 pots destination-pattern 911 port 1 1 1 no register el64 To use a gatekeeper, the user must complete the following three tasks on the gateway 1. Enable the gateway with the gateway...

Congestion Management

Congestion management uses the marking on each packet to determine which queue to place packets in Congestion management utilizes sophisticated queuing technologies such as Weighted Fair Queuing (WFQ) and Low Latency Queuing (LLQ) to ensure that time-sensitive packets like voice are transmitted first 2005 Cisco Systems, Inc. All rights reserved. Congestion management mechanisms (queuing algorithms) use the marking on each packet to determine which queue to place packets in. Different queues are...

Connecting the IP Phone

802.1Q trunking between the switch and IP phone for multiple VLAN support (separation of voice data traffic) is preferred The 802.1Q header contains the VLAN information and the CoS 3-bit field, which determines the priority of the packet 802.1Q trunking between the switch and IP phone for multiple VLAN support (separation of voice data traffic) is preferred The 802.1Q header contains the VLAN information and the CoS 3-bit field, which determines the priority of the packet For most Cisco IP...

Create digital voice ports with the ds0group command

You must create a digital voice port in the T1 or E1 controller to make the digital voice port available for specific voice port configuration parameters. You must also assign timeslots and signaling to the logical voice port through configuration. The first step is to create the T1 or E1 digital voice port with the ds0-group ds0-group-no timeslots timeslot-list type signal-type command. The ds0-group command automatically creates a logical voice port that is numbered as The dsO-group-no...

CRTP Packet Components

In a packet voice environment when speech samples are framed every 20 ms, a payload of 20 bytes is generated. Without CRTP, the total packet size includes the following components The header is twice the size of the payload IP UDP RTP (20 + 8 + 12 40 bytes) versus payload (20 bytes). When generating packets every 20 ms on a slow link, the header consumes a large portion of bandwidth. In the figure, RTP header compression reduces the header to 2 bytes. The compressed header is one tenth of the...

Default Dial Peer

All rights re 2005 Cisco Systems, Inc. All rights re When a matching inbound dial peer is not found, the router resorts to the default dial peer. Note Default dial peers are used for inbound matches only. They are not used to match outbound calls that do not have a dial peer configured. The default dial peer is referred to as dial peer 0.

Default Dial Peer 0

All rights re 2005 Cisco Systems, Inc. All rights re When determining how inbound dial peers are matched on a router, it is important to note whether the inbound call leg is matched to a POTS or VoIP dial peer. Matching occurs in the following manner Inbound POTS dial peers are associated with the incoming POTS call legs of the originating router or gateway. Inbound VoIP dial peers are associated with the incoming VoIP call legs of the terminating router or gateway....

Define the customer administrator credentials

In the Cisco CME system there is a system administrator that has full control of the system. It may be desirable to create another custom level of access to the system by configuring what is known as a customer administrator. This customer administrator can have a subset of the full level of access enjoyed by the default system administrator. The end result will be the existence of two levels of administrators one with full access and the customer administrator with some defined subset of full...

Delay Budget

Delay is the accumulated latency of end-to-end voice traffic in a VoIP network. The purpose of a delay budget is to ensure that the voice network does not exceed accepted limits of delay for voice telephony conversation. The delay budget is the sum of all the delays, fixed and variable, that are found in the network along the audio path. You can measure the delay budget by adding up all of the individual contributing components, as shown in the figure. The delay budget is measured in each...

Delay variation jitter

The clarity, or cleanliness and crispness, of the audio signal is of utmost importance. The listener should recognize the identity of the speaker and sense the mood. Factors that can affect clarity include Fidelity Consistency of transmission bandwidth to the original bandwidth. The bandwidth of the transmission medium almost always limits the total bandwidth of the spoken voice. Human speech typically requires a bandwidth from 100 to 10,000 Hz, although 90 percent of speech intelligence is...

Dialplanpattern tag pattern extensionlength length extensionpattern pattern

The following commands will need to be configured to deploy a Cisco CME system. load phone-type firmware-file ip source-address ip-address port port create cnf-files keepalive seconds dialplan-pattern tag pattern extension-length length extension-pattern pattern In the addition to these commands, ephones and ephone-dns will need to be manually configured. These topics were discussed in a previous lesson.

Dictionaryxml SCCPdictionaryxml

Contents will vary based upon language selected with the user-locale command 1.0 encoding ISO-8859-1 > t Incompatible device type > t Another Barge exists > t Failed to setup Barge > t Network congestion,rerouting > t Not Enough Bandwidth > The files SCCP-dictionary.xml and phonemodel-dictionary.xml configure the language for the IP phones in the system. These contain the labels for buttons as well as messages that could be displayed on the screen of the IP phones. This is set with...

Digit Collection

The router collects digits, one at a time, until it can match an outbound dial peer. After a match is made, the router immediately places the call. No further digits are collected. Example 1 - dialed string is 5550124 Example 2 - dialed string is 5550124 dial-peer voice 1 voip destination-pattern 555 session target ipv4 10.18.0.1 dial-pear voice 2 voip destination-pattern 5550124 session target ipv4 10.18.0.2 dial-peer voice 1 voip destination-pattern 555 session target ipv4 10.18.0.1 dial-peer...

Digit Consumption and Forwarding

POTS dial peers - by default the router consumes the left-justified digits that explicitly match the destination pattern and forwards wiidcarded digits POTS dial peers - use the no digit-strip command to disable the automatic digit-strippingfunction VoIP dial peers - by default the router forwards all digits collected Example 1 - dialed digits 5551234 Example 2 - dialed digits 5551234 dial-peer voice X pots deetination-pattern BSE . port 1 0 1 diel-peer voice 1 pots destination-pattern 555,,, ....

Digit Manipulation Commands

Adds digits to the front of the dial string before it is forwarded to the telephony interface Controls the number of digits forwarded to the telephony interface Expands an extension into a full telephone number or replaces one number with another Digit translation rules used to manipulate the calling number digits, or ANI, or the called number digits, or DNIS, for a voice call Digit manipulation is the task of adding or subtracting digits from the original dialed number to accommodate user...

Digit Maps

A digit map is a specification of the dial plan. When you download a digit map to a gateway for use on an endpoint or a group of endpoints, a digit map allows the gateway to collect digits until the gateway either finds a match or concludes that the dialed digits could not possibly match the specification. When either condition occurs, the gateway notifies the call agent. Without a digit map, a gateway must notify the call agent on each digit dialed, which places a heavy burden on the call...

Directs calling side to seize the Elead and send DTMF digits

The E& M port that music on hold arrives on will also need to be configured in 4-wire mode. This is done by entering the operation 4-wire command. E& M ports also needs to be configured to proceed with connecting the call by seizing the line and sending DTMF digits without waiting for any signal from the other side of the connection. This is done by the use of the command signal immediate. Example Router(config-voice-port) operations 4-wire (E& M ports only) Selects the 4-wire cabling...

Distributed Call Control

This figure shows an environment where call control is handled by multiple components in the network. Distributed call control is possible where the voice-capable device is configured to support call control directly. This is the case with a voice gateway when protocols, such as H.323 or SIP, are enabled on the device. Distributed call control enables the gateway to perform the following procedure I. Recognize the request for service

Dn

Ring no answer timeout of 10 seconds set globally Ring no answer timeout of 10 seconds set globally When the no huntstop command is used on the ephone-dn, the call would ring on the first ephone-dn and go through any hunting defined on the two channels in a dual-line ephone-dn before being sent to the next most preferred ephone-dn that also has a matching destination pattern. This will continue until an ephone-dn with huntstop configured is reached or no more dial peers (ephone-dns) have...

Dscp

The newest use of the eight CoS bits is commonly called the DiffServ standard. It uses the same precedence bits (the most significant bits 1, 2, and 3) for priority setting, but further clarifies their functions and definitions and offers finer priority granularity through use of the next three bits in the CoS field. DiffServ reorganizes and renames the precedence levels (still defined by the three most significant bits of the CoS field) into the categories shown in the table. Stays the same...

Dynamic Mode

The playout-delay command allows you to select a jitter buffer mode (static or dynamic) and specify certain values that are used by the DSP algorithms to adjust the size of the jitter buffer. During a voice call, the algorithms read time stamps in the RTP headers of sample packets to determine the amount of delay that the jitter buffer will apply to an average packet that is, as if there is no jitter at all in the network. This is called the average delay.

Echo Canceller Comparison

This table contains echo canceller comparison information. Configurable to greater than or equal to -0 dB, -3 dB, or -6 dB Not required due to faster convergence 12.2(11 )T, 12.2(8)T5, 12.2(12), and higher 12.2(13)T, 12.2(8)YN, 12.2(15)T, 12.3(4)T, 12.3(4)XD, and higher

Echo Is Always Present

Echo as a problem is a function of the echo delay and the loudness of the echo. Some form of echo is always present. However, echo can become a problem under the following conditions The magnitude or loudness of the echo is high. The delay time between when you speak and when you hear your voice reflected is significant. The listener hears the speaker twice. The two components of echo are loudness and delay. Reducing either component reduces overall echo. When a user experiences delay, the...

Echo Suppression

The echo suppressor works by transmitting speech in the forward direction and prohibiting audio in the return direction. The echo suppressor essentially breaks the return transmission path. This solution works sufficiently for voice transmission. However, for full-duplex modem connections, the action of the echo suppressor prevents communication. Therefore, when modems handshake, the answering modem returns a tone of 2025 Hz to the calling modem, which serves to disable the echo suppressors...

Edge Devices

The two types of edge devices that are used in a telephony network include Analog telephones Analog telephones are most common in home, small office home office (SOHO), and small business environments. Direct connection to the PSTN is usually made by using analog telephones. Proprietary analog telephones are occasionally used in conjunction with a PBX. These telephones provide additional functions such as speakerphone, volume control, PBX message-waiting indicator, call on hold, and...

Enables call completion when no Mlead response is sent

If the MoH arrives on an analog port that FXO or E& M port will need to be configured. The command input gain allows the volume of the feed to be tuned up or down on either an FXO or E& M port. E& M ports require additional configuration, one of these E& M command is the auto-cut-through command which allows the connection to the feed to be set up even though the source of the feed will not provide a M-lead response. Example Router(config) voice-port 1 0 1 Enters voice port...