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...

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.

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...

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...

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...

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...

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 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...

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 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)

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...

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...

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 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

Member 911 dial-peer cor list callLocal member local dial-peer cor list callLD member long_distance dial-peer cor list callInt member international dial-peer cor list call900 member 900 dial-peer cor list Lobby member 911 dial-peer cor list Employee member 911 member local dial-peer cor list Sales member 911 member local member long_distance dial-peer cor list Executive member 911 member local member long_distance member international

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

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...

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...

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...

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....

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...

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...

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...

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...

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...

Endto End Calls

An end-to-end voice call consists of four call legs two from the originating router (R1) or gateway perspective, and two from the terminating router (R2) or gateway perspective. An inbound call leg originates when an incoming call comes into the router or gateway. An outbound call leg originates when a call is placed from the router or gateway. A call is segmented into call legs and a dial peer is associated with each call leg. The process for call setup is listed below 1. The plain old...

Ephone

Software configuration of a physical phone Has a unique tag or sequence number assigned when the ephone is created Can be an IP phone, analog phone attached to an ATA The MAC of the IP phone or ATA is used to tie the software configuration to the hardware The hardware is auto detected for all supported models except the ATA and 7914 expansion module Can have one or more ephone-dn(s) associated with the ephone Number of line buttons will vary based on the hardware Button 1) DN Button 4J Button 2...

Ephone Cont Basic Example

CMERouter(Config) ephone-dn 7 CMERouter(Config-ephone-dn) number 1001 CMERouter(config) ephone 1 CMERouter(config-ephone) mac-address 000F.2470.F8F8 CMERouter(config-ephone) button 1 7 2005 Cisco Systems, Inc. All rights re This example shows an ephone-dn 7 being created and then assigned to the ephone 1. The ephone-dn is configured to be dual-line and is assigned to line button one on the IP phone at the specified MAC address. Ephone (Cont.) Example Multiple Ephones When there are multiple...

Ephone Cont Example Multiple Ephones Configuration

CMERouter(config) ephone-dn 10 dual-line CMERouter(config-ephone-dn) number 1004 CMERouter(config) ephone-dn 11 dual-line CMERouter(config-ephone-dn) number 1005 CMERouter(config) ephone-dn 12dual-line CMERouter(config-ephone-dn) number 1006 CMERouter(config) ephone-dn 13 dual-line CMERouter(config-ephone-dn) number 1007 CMERouter(config) ephone 1 CMERouter(config-ephone) mac-address 000F.2470.F8F1 CMERouter(config-ephone) button 1 10 CMERouter(config) ephone 2 CMERouter(config-ephone)...

Ephonedn

An ephone-dn, or Ethernet phone directory number, is a software construct that represents the line that connects a voice channel to a phone instrument on which a user can receive and make calls. An ephone-dn has one or more extension or telephone numbers associated with it to allow call connections to be made. An ephone-dn is equivalent to a phone line in most cases, but not always. There are several types of ephone-dns, which have different characteristics. Each ephone-dn has a unique dn-tag,...

Example Apply the COR to the dial peer

CMERouter(config) dial-peer voice 1 pots 1500 CMERouter(config-dial-peer) port 1 0 0 CMERouter(config-dial-peer) corlist incoming call911 CMERouter(config) dial-peer voice 2pots 1800 CMERouter(config-dial-peer) port 2 1 CMERouter(config-dial-peer) corlist outgoing call1800 Steps to Configure Class of Restriction Step 4 - Assign the COR list to the ephone-dns Defines an ephone-dn and enters ephone-dn mode Specifies a COR list to be used when the ephone-dn is used as either the incoming or...

Example Channel Associated Signaling

D4 has a 12-frame structure and provides AB bits for signaling. ESF has a 24-frame structure and provides ABCD bits for signaling. DTMF, or tone, can be carried in band in the audio path however, other supervisory signals must still be carried via CAS. This topic describes CAS and its uses with E1 transmission. In E1 framing and signaling, 30 of the 32 available channels, or timeslots, are used for voice and data. Framing information uses timeslot 1, while timeslot 17 (E0 16) is used for...

Example Cisco Reliability and Availability

In some data networks, a high level of availability and reliability is not critical enough to warrant financing the hardware and links required to provide complete redundancy. If voice is layered onto the network, these requirements need to be revisited. With Cisco Architecture for Voice, Video and Integrated Data (AVVID) technology, the use of Cisco CallManager clusters provides a way to design redundant hardware in the event of Cisco CallManager failure. When using gatekeepers, you can...

Example Configuring FXS Ports

For example, consider the scenario of an international company with offices in the United States and England. The PSTN of each country provides signaling that is standard for that country. In the United States, the PSTN provides a dial tone that is different from the tone in England. When the telephone rings to signal an incoming call, the ring is different in the United States. Another instance when the default configuration might be changed is when the connection is a trunk to a PBX or key...

Example COR naming and list

CMERouter(config) dial-peer cor custom CMERouter(config-dp-cor) name local_call CMERouter(config-dp-cor) name 911 CMERouter(config-dp-cor) name 1800 CMERouter(config-dp-cor) name 1900 Step 2 Dial peer COR list and member commands set the capabilities of a COR list. COR list is used in dial peers to indicate the restriction that a dial peer has as an outgoing dial peer. The order of entering the members is not important and the list can be appended or made shorter by removing the members.

Example COR used to restrict access internally within Cisco CME

COR can be used to regulate internal calls and whether they are allowed or not. This example shows two IP phones with an employee and an executive. In this company, the executive should be able to call anyone but employees should not be able to call the executive. Notice that to accomplish the required results, both an incoming COR on the employee must be configured as well as an outgoing COR on the executive. There is no outgoing COR on the employee and as a result anyone can call the employee...

Example Decibel Levels

Adjustment of decibel levels may be necessary throughout a voice network. A station connected to a PBX may experience one level of loudness when calling a local extension, a different level when dialing an outside line, and different levels when calling remote sites via VoIP. Adjustments may be necessary in this case. This topic describes voice-quality tuning configuration. Configuring Voice Port Voice-Quality Tuning

Example Define the COR lists

CMERouter(config) dial-peer list callLocal CMERouter(config-dp-corlist)member local_call CMERouter(config) dial-peer list call911 CMERouter(config-dp-corlist)member 911 CMERouter(config) dial-peer list call1800 CMERouter(config-dp-corlist)member 1800 CMERouter(config) dial-peer list call1900 CMERouter(config-dp-corlist)member 1900 This is the third step to configure Class of Restriction (COR). Steps to Configure Class of Restriction Step 3 - Assign the COR list to the dial peers dial-peer voice...

Example Dial Peer Configuration

This figure shows a dial-peer configuration. In the figure, the telephony device connects to the Cisco Systems voice-enabled router. The POTS dial-peer configuration includes the telephone number of the telephony device and the voice port to which it is attached. The router knows where to forward incoming calls for that telephone number. The Cisco voice-enabled router VoIP dial peer is connected to the packet network. The VoIP dial-peer configuration includes the destination telephone number...

Example E1 Channel Associated Signaling

E1 CAS is directly compatible with T1 CAS, because both methods use AB or ABCD bit signaling. Although the signaling for E1 CAS is carried in a single common timeslot, it is still referred to as CAS because each individual signaling timeslot represents a specific pair of voice channels. This topic describes common channel signaling (CCS) systems. CCS differs from CAS in that all channels use a common channel and protocol for call setup. Using E1 as an example, a signaling protocol, such as the...

Example Fragment Size Configuration

You can configure fragment size using the frame-relay fragment fragmentsize command in a Frame Relay map class. The fragmentsize argument defines the payload size of a fragment and excludes the Frame Relay headers and any Frame Relay fragmentation header. The valid range is from 16 bytes to 1600 bytes the default is 53 bytes. The fragment size argument should be set so that the serialization delay is close to 10 ms for example, if using a 384-kbps link, the fragmentation size should be set at...

Example FXO Port Configuration

The configuration in the figure enables loop-start signaling on a Cisco 2600 or 3600 series router on FXO voice port 1 0 0. The ring-number setting of 3 specifies that the FXO port does not answer the call until after the third ring, and the dial type is set to DTMF. Voice-port configuration on voice-enabled router Voice-port configuration on voice-enabled router Enters voice-port configuration mode Enters voice-port configuration mode

Example Matching Destination Patterns

Matches one telephone number exactly, 5550124. This is typically used when there is a single device, such as a telephone or fax, connected to a voice port. Matches a seven-digit telephone number where the first five digits are 55501, the sixth digit can be a 1, 2, or 3, and the last digit can be any valid digit. This type of destination pattern is used when telephone number ranges are assigned to specific sites. In this example, the destination pattern is used in a small site that does not need...

Example MGCP Residential Gateway Configuration

In the example, the configuration identifies the packages that the gateway expects the call agent to use when it communicates with the gateway. The last mgcp command specifies the default the gateway uses if the call agent does not share the capabilities. In this example, the command is redundant because the line package is the default for a residential gateway. When the parameters of the MGCP gateway are configured, the active voice ports (endpoints) are associated with the MGCP. Dial peer 1...

Example Need For a Delay Budget

As delay increases, talkers and listeners become unsynchronized and often find themselves speaking at the same time or both waiting for the other to speak. This condition is commonly called talker overlap. While the overall voice quality may be acceptable, users may find the stilted nature of the conversation unacceptably annoying. Talker overlap may be observed on international telephone calls that travel over satellite connections. Satellite delay is about 500 ms 250 ms up and 250 ms down.

Example Number Normalization

When site E (703555 ) dials 7275550199, the full 10-digit dialed string is passed through the Centrex to the router at site D. Router D matches the destination pattern 7275550199 and forwards the 10-digit dial string to router A. Router A matches the destination pattern 727555 , strips off the matching 727555, and forwards the remaining 4-digit dial string to the PBX. The PBX matches the correct station and completes the call to the proper extension. Calls in the reverse direction are handled...

Example QoS Objectives

VoIP guarantees high-quality voice transmission only if the signaling and audio channel packets have priority over other kinds of network traffic. To deploy VoIP, you must provide an acceptable level of voice quality by meeting VoIP traffic requirements for issues related to bandwidth, latency, and jitter. QoS provides better, more predictable network service by performing the following Supporting dedicated bandwidth Designing the network such that speeds and feeds can support the desired voice...

Example Reordering Voice Packets

In the figure, RTP reorders the voice packets through the use of sequence numbers before playing them out to the user. The table illustrates the various stages of packet reordering by RTP. IP assumes packet-ordering problems. The voice packets are put in order through the use of sequence numbers. The voice packets are spaced according to the time stamp contained in each RTP header. The user hears the voice packets in order and with the same timing as when the voice stream left the source. RTCP...

Example Technology Prefixes Applied

Voice gateways can register with technology prefix 1 H.320 gateways with technology prefix 2 and voice-mail gateways with technology prefix 3 . Multiple gateways can register with the same type prefix. When this happens, the gatekeeper makes a random selection among gateways of the same type. If the callers know the type of device that they are trying to reach, they can include the technology prefix in the destination address to indicate the type of gateway to use to get to the destination. For...

Example Using the Port Specific AutoQoS Macro

This example shows how to use the ciscoipphone keyword Console> (enable) set port qos 3 1 autoqos help Usage set port qos < mod port> autoqos trust < cos dscp> set port qos < mod port> autoqos voip < ciscoipphone ciscosoftphone> Console> (enable) set port qos 3 1 autoqos voip ciscoipphone Port 3 1 ingress QoS configured for Cisco IP Phone. It is recommended to execute the set qos autoqos global command if not executed previously. Console> (enable) This example shows how to...

Example Voice Port Applications

The table lists application examples for each type of call. One staff member calls another staff member at the same office. The call is switched between two ports on the same voice-enabled router. One staff member calls another staff member at a remote office. The call is sent from the local voice-enabled router, across the IP network, and terminated on the remote office voice-enabled router. A staff member calls a client who is located in the same city. The call is sent from the local...

Example Waveform Compression

Differential encode changes between samples only Standard PCM is known as ITU standard G.711. Adaptive differential pulse code modulation (ADPCM) coders, like other waveform coders, encode analog voice signals into digital signals to adaptively predict future encodings by looking at the immediate past. The adaptive feature of ADPCM reduces the number of bits per second that the PCM method requires to encode voice signals. ADPCM does this by taking 8000 samples per second of the analog voice...

Example

The following is a sample configuration using the prefix command In the sample configuration using the prefix command, the device attached to port 1 0 0 needs all seven digits to process the call. On a POTS dial peer, only wildcard-matched digits are forwarded by default. Use the prefix command to send the prefix numbers of 555 before forwarding the four wildcard-matched digits. The following is a sample configuration using the forward-digits command In the sample configuration using the...

Filenames are casesensitive

All rights re To associate a type of Cisco IP phone with a phone firmware file, use the load model firmware-file command in telephony-service configuration mode. The following show the supported models for which firmware can be loaded Note No suffix should be used when using the load command for the 7910, 7940 and 7960 model 7902 Select the firmware load file for 7902 7905 Select the firmware load file for 7905 7910 Select the IP phone firmware load file for Telecaster...

Files Cont Call Progress XML File

< tone c1 30831 i1 -2032 c2 30467 i2 -1104 d 2 t ringing> < part m on t 2000 > < part m off t 4000 > < repeat c 65535 > < tone> < tone c1 30467 i1 -1104 c2 28959 i2 -1404 d 2 t reorder> < part m on t 250 > < part m off t 250 > < repeat c 65535 > < tone> < tone c1 30467 i1 -1104 c2 28959 i2 -1404 d 2 t busy> < part m on t 500 > < part m off t 500 > < repeat c 65535 > < tone> < tone c1 30743 i1 -1384 c2 29780 i2 -1252 d 2 t odial>...

Foreign Exchange Station

Connects directly to station equipment Used to provision local service Foreign exchange (FX) trunks are interfaces that are connected to switches that support connection to either office equipment or station equipment. Office equipment includes other switches (to extend the connection) and Cisco devices. Station equipment includes telephones, fax machines, and modems. Foreign Exchange Office (FXO) interfaces An FXO interface connects a PBX to another switch or Cisco device. The purpose of an...

Foreign exchange station interface

Connects directly to analog phones or faxes Used to provision local service Provides power, call progress tones, and dial tone When analog phones or fax machines are used in an IP-based environment, they will need to have a connection into this IP network. This connection takes the form of an FXS interface. The FXS interface provides a direct connection to an analog telephone, a fax machine, or a similar device. From the analog device's perspective, the FXS interface functions like a switch....