Modem relay demodulates a modem signal at one voice gateway and passes it as packet data to another voice gateway, where the signal is remodulated and sent to a receiving modem. On detection of the modem answer tone, the gateways switch into modem pass-through mode and then, if the call menu CM signal is detected, the two gateways switch into modem relay mode. Although modem pass-through remains susceptible to packet loss, jitter, and latency in the IP network, packet redundancy can be used to mitigate the effects of packet loss in the IP network.
SPRT is a protocol running over UDP packets to the other gateway, where the modem signal is re-created, remodulated, and passed to the receiving modem. In this implementation, the call starts out as a voice call, switches into modem passthrough mode, and then into modem relay mode. Modem relay significantly reduces the effects that dropped packets, latency, and jitter have on the modem session. Compared to modem pass-through, it also reduces the amount of bandwidth used.
It forces higher-rate modems to train down to the supported rates. You can configure the codec by using the galaw or gulaw option of the codec command. Payload Redundancy You can enable payload redundancy so the modem pass-through over VoIP switchover causes the gateway to send redundant packets.
Redundancy can be enabled in one or both of the gateways. When only a single gateway is configured for redundancy, the Chapter 2: Considering VoIP Design Elements other gateway receives the packets correctly, but does not produce redundant packets. When redundancy is enabled, 10 ms sample-sized packets are sent. When redundancy is disabled, 20 ms sample-sized packets are sent.
Note By default, the modem relay over VoIP capability and redundancy are disabled. Dynamic and Static Jitter Buffers When gateways detect a data modem, both the originating gateway and the terminating gateway switch from dynamic jitter buffers to static jitter buffers of ms depth.
The switch from dynamic to static is designed to compensate for PSTN clocking differences at the originating and terminating gateways. When the modem call is concluded, the voice ports revert to dynamic jitter buffers.
This new feature is a non-negotiated, bearer-switched mode for modem transport that does not involve call-agent-assisted negotiation during the call setup. Instead, the negotiation parameters are configured directly on the gateway. These gateway-controlled negotiation parameters use NSEs to indicate the switchover from voice to voice band data to modem relay.
Upon detecting a Hz tone, the terminating gateway sends an NSE to the originating gateway and switches over to modem pass-through. The terminating gateway also sends an NSE to indicate modem relay. If this event is recognized by the originating gateway, the call occurs as modem relay. If the event is not recognized, the call occurs as modem pass-through. Because Cisco modem relay uses configured parameters, it removes the signaling dependency from the call agent and allows modem relay support independent of call control.
The gateway-controlled modem relay parameters are enabled by default when Cisco modem relay is configured.
Interestingly, when Cisco modem relay is configured, gateway XID parameter negotiation is always enabled. The fax e-mail message and attachment are handled by an e-mail server while traversing the packet network and can be stored for later delivery or delivered immediately to a PC or to an off-ramp gateway. On-ramp and off-ramp faxing processes can be combined on a single gateway, or they can occur on separate gateways. Store-and-forward fax uses two different interactive voice response IVR applications for on-ramp and off-ramp functionality.
When a terminating gateway TGW detects a called terminal identification CED tone from a called fax machine, the TGW exchanges the voice codec that was negotiated during the voice call setup for a G. This switchover is communicated to the originating gateway OGW , which allows the fax machines to transfer modem signals as though they were traversing the PSTN. If the voice codec that was configured and negotiated for the VoIP call is G.
If the CED tone is heard, an internal event is generated to alert the call control stack that a fax or modem changeover is required. The fax machines are able to communicate on an end-to-end basis with no further intervention by the voice gateways.
NSEs are a Cisco-proprietary version of IETF-standard named telephony events NTEs , which are specially marked data packets used to digitally convey telephony signaling tones and events. NSEs and NTEs provide a more reliable way to communicate tones and events using a single packet rather than a series of in-band packets that can be corrupted or partially lost.
Fax pass-through and fax pass-through with up speed use peer-to-peer NSEs within the RTP stream or bearer stream to coordinate codec switchover and the disabling of echo cancellation and VAD.
Redundant packets can be sent to improve reliability when the probability of packet loss is high. When a DSP is put into voice mode at the beginning of a VoIP call, the DSP is informed by the call control stack whether or not the control protocol can support pass-through. If Cisco Fax Relay is supported, the following events occur: 1. Initially, a VoIP call is established as if it were a normal speech call.
Call control procedures are followed, and the DSP is put into voice mode, after which human speech is expected to be received and processed. The ANSam or CED tone causes a switch to modem pass-through, if enabled, to allow the tone to pass cleanly to the remote fax. The DSP also triggers an internal event to notify the call control stack that fax switchover is required. When the DSP on the OGW receives an RTP packet with the payload type set to 96, it triggers an event to inform its own call control stack that a fax changeover has been requested by the remote gateway.
When the TGW receives the payload type 97 packet, the packet serves as an acknowledgement. The TGW starts the fax codec download and is ready for fax relay. As part of the fax codec download, other parameters such as VAD, jitter buffers, and echo cancellation are changed to suit the different characteristics of a fax call.
During fax relay operation, the T. The T. The TGW decodes the data stream and remodulates the T. The messages that are demodulated and remodulated are predominantly the phase B, phase D, and phase E messages of a T.
Most of the messages are passed across without any interference, but certain messages are modified according to the constraints of the VoIP network. They expect to communicate with each other across a 64 kbps PSTN circuit, and they attempt to make best use of the available bandwidth and circuit quality of a 64 kbps voice path.
The bandwidth per call is probably less than 64 kbps, and the circuit is not considered a clear circuit. The adjusted fax settings restrict the facilities that are available to fax machines across the VoIP call leg and are also used to modify values in DIS and NSF messages that are received from fax machines.
The encoding of the packet headers and the mechanism to switch from VoIP mode to fax relay mode are clearly defined in the specification. Annexes to the basic specification include details for operation under SIP and H. At the end of the fax transmission, either gateway can initiate another ModeRequest message to return to VoIP mode. During T.
The level of redundancy the number of times the packet is repeated can be configured on Cisco IOS gateways. The capability to support T. If the DSP on the gateway is capable of supporting T.
When a fax tone is heard, the DSP signals the receipt of the fax tone to the call control layer, which then initiates fax changeover as specified in the T. SIP T. When the call control protocol is SIP, T.
G3 Fax Initiates the Call T. The DSP needs to be informed that it can support T. The SIP T. The TGW detects a fax V. The OGW starts sending T. MGCP T. MGCP-based T. A call is initially established as a voice call. The gateways advertise capabilities in an SDP exchange during connection establishment. If both gateways do not support T. If both gateways support T. The existing audio channel is used for T. If failure occurs at some point during the switch to T. If this failure occurs, a fallback to fax pass-through is not supported.
Upon completion of the fax image transfer, the connection remains established and reverts to a voice call using the previously designated codec, unless the call agent instructs the gateways to do otherwise. A fax relay MGCP event allows the gateway to notify the call agent of the status start, stop, or failure of T.
This event is sent in both call agent-controlled and gateway-controlled mode. Gateways do not need instruction from the call agent to switch to T. This mode is used if the call agent has not been upgraded to support T. Gateway-controlled mode can also be used to bypass the message delay overhead caused by call agent handling for example, to meet time requirements for switchover to T.
If the call agent does not specify the mode to the gateway, the gateway defaults to gateway-controlled mode. In gateway-controlled mode, the gateways exchange NSEs by performing these steps: 1. Instruct the peer gateway to switch to T.
Either acknowledge the switch and the readiness of the gateway to accept T. This new capability allows successful T. New ports are assigned in H. Gateways send these tones in the RTP stream by default. This default behavior is fine when the voice stream is sent uncompressed, but problems arise when sending voice across slower WAN links using compression algorithms, as illustrated in Figure As a result, IVR systems might not correctly recognize the tones.
However, the DTMF tones are encoded differently from the voice samples and are identified as payload type , which enables the receiver to identify them as DTMF tones.
This method requires the use of Cisco gateways at both the originating and terminating endpoints of the H. The tones are transported in H. This method does not send tone length information. This method is optional on H. Note All H. RFC support is standards-based and allows greater interoperability with other gateways and call agents. That transmission is transparent to the call agent. Gateway-controlled mode allows the use of the DTMF relay feature without upgrading the call agent software to support the feature.
Out-of-band: Sends the tones as signals to Cisco Unified Communications Manager out-of-band over the control channel. Cisco Unified Communications Manager interprets the signals and passes them on.
The terminating gateway acknowledges the message with an 18x or Response message, also using the Call-Info header. In response, the gateway expects to receive a OK message. Processing Voice Packets with Codecs and DSPs Because WAN bandwidth is probably the most expensive component of an enterprise network, network administrators must know how to calculate the total bandwidth required for voice traffic and how to reduce overall bandwidth consumption. Several variables affecting total bandwidth are explained, as well as how to calculate and reduce total bandwidth consumption.
Chapter 2: Considering VoIP Design Elements Codecs A codec is a device or program capable of performing encoding and decoding on a digital data stream or signal. Various types of codecs are used to encode and decode or compress and decompress data that would otherwise use large amounts of bandwidth on WAN links. Codecs are especially important on lower-speed serial links where every bit of bandwidth is needed and utilized to ensure network reliability. One of the most important factors for a network administrator to consider while building voice networks is proper capacity planning.
Network administrators must understand how much bandwidth is used for each VoIP call. To understand bandwidth, the administrator must know which codec is being utilized across the WAN link.
With a thorough understanding of VoIP bandwidth and codecs, the network administrator can apply capacity planning tools. Coding techniques are standardized by the ITU. It is a PCM scheme operating at an 8 kHz sample rate, with 8 bits per sample. With G. It is widely used in the telecommunications field because it improves the signal-to-noise ratio without increasing the amount of data.
There are two subsets of the G. Both mu-law and a-law subsets use digitized speech carried in 8-bit samples. They use an 8 kHz sampling rate with 64 kbps of bandwidth demand. The four bit rates associated with G. Also, G. The features of G. Standard G. This compression technique can be used for compressing speech or audio signal components at a very low bit rate as part of the H.
The lower bit rate is based on CELP and provides system designers with additional flexibility. The network messaging must be capable of recording a voice message and depositing the message to an external server for later retrieval.
This codec supports the Cisco infrastructure and application partner components required for service providers to deploy unified messaging applications. The algorithm is a version of Block-Independent Linear Predictive Coding, with the choice of data frame lengths of 20 and 30 milliseconds.
The encoded blocks have to be encapsulated in a suitable protocol for transport, such as RTP. This codec enables graceful speech quality degradation in the case of lost frames, which occurs in connection with lost or delayed IP packets. The network administrator should balance the need for voice quality against the cost of bandwidth in the network when choosing codecs.
The higher the codec bandwidth, the higher the cost of each call across the network. 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 by default, regardless of the codec used. You can apply an optional configuration command to vary the number of samples encapsulated. When you encapsulate more samples per PDU, the total bandwidth is reduced. Table demonstrates how the number of packets required to transmit one second of audio varies with voice sample sizes.
Depending on the Layer 2 protocol used, the overhead could grow substantially. More bandwidth is required to transport VoIP frames with larger Layer 2 overhead. When using a virtual private network VPN , IP Security IPsec will add 50 to 57 bytes of overhead, a significant amount when considering the relatively small voice-packet size. When using MLP, 6 bytes will be added to each packet.
All these specialized tunneling and security protocols must be considered when planning for bandwidth demands. For example, many companies have their employees telecommute from home. These employees often initiate a VPN connection into their enterprise for secure Internet transmission. Calculating the Total Bandwidth for a VoIP Call Codec choice, data-link overhead, sample size, and RTP header compression have positive and negative impacts on total bandwidth, as demonstrated in Table Consider a sample total bandwidth calculation.
A company is implementing VoIP to carry voice calls between all sites. WAN connections between sites will carry both data and voice. To use bandwidth efficiently and keep costs to a minimum, voice traffic traversing the WAN will be compressed using the G.
WAN connectivity will be through a Frame Relay provider. With traditional telephony voice networks, all G. In Cisco VoIP networks, all conversations and silences are packetized. VAD can suppress packets containing silence. Table illustrates the type of bandwidth savings VAD offers. Note For the purposes of network design and bandwidth engineering, VAD should not be taken into account, especially on links that carry fewer than 24 voice calls simultaneously.
When the network is engineered for the full voice-call bandwidth, all savings provided by VAD are available to data applications. In some cases, silence might be mistaken for a disconnected call.
CNG provides locally generated white noise to make the call appear normally connected to both parties. The company plans to use G. Previously, it was determined that each voice call compressed with G. VAD can reduce the bandwidth utilization to approximately 17, bps, which constitutes a bandwidth savings of 35 percent. DSPs enable Cisco platforms to efficiently process digital voice traffic.
DSPs on a router provide stream-to-packet signal processing functionality that includes voice compression, echo cancellation, and tone- and voice-activity detection.
A media resource is a software-based or hardware-based entity that performs mediaprocessing functions on the data streams to which it is connected. Transcoding compresses and decompresses voice streams to match endpoint-device capabilities. Transcoding is required when an incoming voice stream is digitized and compressed by means of a codec to save bandwidth, but the local device does not support that type of compression.
Ideally, all IP telephony devices would support the same codecs, but this is not the case. Rather, different devices support different codecs. Sessions are initiated and managed by Cisco Unified Communications Manager. If an application or service can handle only one specific codec type, which is usually G. This can be done only via DSP resources.
Because applications and services are often hosted in main sites, DSP transcoding resources are most common in central sites. DSPs perform this termination function. The DSP also provides echo cancellation, voice activity detection, and jitter management at the same time it performs voice termination. It bridges the media streams and allows them to be set up and torn down independently. The streaming data received from the input stream on one connection is passed to the output stream on the other connection, and vice versa.
In addition, the MTP can be used to transcode a-law to mu-law and vice versa, or it can be used to bridge two connections that utilize different packetization periods. Because IP phones transmit voice traffic directly between phones, a network-based conference bridge is required to facilitate multiparty conferences. It can accept any number of connections for a given conference, up to the maximum number of streams allowed for a single conference on that device.
A one-to-one correspondence exists between media streams connected to a conference and participants connected to the conference. The conference bridge mixes the streams together and creates a unique output stream for each connected party. The output stream for a given party is the composite of the streams from all connected parties minus their own input stream.
Some conference bridges mix only the three loudest talkers on the conference and distribute that composite stream to each participant minus their own input stream if they are one of the talkers. Hardware conference bridges are used in two environments. They can be used to increase the conferencing capacity in a central site without putting an additional load on Cisco Unified Communications Manager servers, which can host software-based conference bridges.
More important is the use of hardware conference bridges in remote sites. If no remote-site conference resources are deployed, every conference will be routed to central resources, resulting in sometimes-excessive WAN usage. In addition, DSP-based conference bridges can mix G.
In contrast, software-based conference bridges deployed on Cisco Unified Communications Manager servers can mix only G. This requirement occurs infrequently.
Cisco H. When needed, an MTP is allocated and connected into a call on behalf of an H. After insertion, the media streams are connected between the MTP and the H. The media streams connected to the other side of the MTP can be connected and disconnected as needed to implement features such as hold and transfer. When an MTP is required on an H. A software MTP device supports G.
The IP Voice Media Streaming Application is a resource that might also be used for several functions, and proper design must consider all functions together. However, some of these are not pertinent to a Cisco Unified Communications Manager implementation. The router configuration permits up to individual streams, which support transcoded sessions. This number of G.
This is done without using the gateway CPU. This hardware-only implementation uses a DSP resource for endpoints using the same G. The repacketization requires a DSP resource, so it cannot be done by software only. Eight ports are available per module. The transcoding resources can be used to transcode G.
Codec Complexity Codec complexity refers to the amount of processing required to perform voice compression. Codec complexity affects call density that is, the number of calls reconciled on the DSPs. With higher codec complexity, fewer calls can be handled.
Select a higher codec complexity when that is required to support a particular codec or combination of codecs. Select a lower codec complexity to support the greatest number of voice channels, provided the lower complexity is compatible with the particular codecs in use. Table illustrates the complexity modes the C chipset needs to run to support a variety of codecs. The difference between medium and high complexity codecs is the amount of CPU utilization necessary to process the codec algorithm, and therefore, the number of voice channels that can be supported by a single DSP.
For this reason, all the medium complexity codecs can also be run in high complexity mode, but fewer usually about half of the channels are available per DSP. To specify call density and codec complexity according to the codec standard that is used, use the codec complexity command in voice-card configuration mode. Consider Examples and , which show the supported codec complexity modes for the C and C DSPs, using context-sensitive help. Notice the C DSPs support a flex complexity mode, which allows the DSPs to automatically switch into the optimal complexity mode for a given call, unlike the C DSPs, which require you to use the high complexity mode which supports the fewest number of calls if the DSPs ever need to run in high complexity mode.
High complexity, lower call density. Mid range complexity and call density. Flex complexity, higher call density. Router config-voicecard codec complexity flex Router config-voicecard When you use flex complexity, up to 16 calls can be completed per DSP.
The number of supported calls varies from 6 to 16 and is based on the codec used for a call. The show voice dsp command, as demonstrated in Example , can be used to verify codec complexity configurations. This mainly depends on two factors: DSP type and the codec being used. The configuration specifies which function each DSP will perform. Multiple profiles can be defined on a single gateway. These profiles can then be registered to different Cisco Unified Communications Manager clusters.
This means that whether a conference has three, five, or eight participants, it counts the same against the number of simultaneous conferences supported on a DSP. Table shows the various DSP resources for conferencing and their performance.
C support up to four transcoding sessions for any codec combination. The C supports 16 G. Table shows the various DSP resources that can be used for transcoding and their performance. Select the correct router model, in this case Cisco Step 2. In this case, Step 3. Select the appropriate VIC configuration. Step 4. Specify the maximum number of calls for a specific codec or fax configuration. Figure 4 Specify the number of calls.
Step 5. Specify the number of transcoding sessions with the appropriate codec. In this example, 8 G. Step 6. Specify the number of conferences required on the gateway, either singlemode G. Figure Step 7. The optimized result uses the Cs in flex mode, and the normal result uses either medium or high complexity mode, depending on the used codecs. You should use flex-mode because of higher performance and fewer required DSP resources.
In rare cases, this might lead to oversubscribed DSP resources. The basic steps for configuring conferencing and transcoding on voice gateway routers are as follows: Step 1. You must determine the number of PVDM2s or network modules required to support your conferencing and transcoding services and install the modules on your router.
Therefore, SCCP needs to be enabled and configured on the router. Examples are provided for each configuration task. Cisco Unified Communications Manager requests conferencing or transcoding services from the gateway, which either grants or denies these requests, depending on resource availability.
Consider the topology in Figure Under the profile, you select the service type conference, transcode, MTP , associate an application, and specify service-specific parameters such as codecs and the maximum number of sessions. Applications associated with the profile, such as SCCP, can use the resources allocated under the profile. You can configure multiple profiles for the same service, each of which can register with one Cisco Unified Communications Manager group.
In this example, because transcoding is required on Router1, the dspfarm profile 1 transcoding command is used. On Router2, the dspfarm profile 1 conferencing command creates a profile for conferencing.
Because both G. Configurations for Router1 and Router2 are provided in Examples and If only G. Because Cisco Unified Communications Manager 4. Note that the San Jose Cisco Unified Communications Manager server references the identifier option previously specified. These commands are issued on both gateways, Router1 and Router2, as illustrated in Examples and Continuing with the current example, a conference bridge is defined in the Service, Media Resource, Conference Bridge menu, as shown in Figure Figure Navigating to the Conference Bridge Configuration Screen The newly added conference bridge now needs to be set up.
After you select the correct type, specify the parameters described in Table and illustrated in Figure Device Pool Default Select the correct device pool.
Location Select the correct location. Note For simplicity, the device pool and location are left at their defaults. After you select the correct type, specify the parameters as described in Table and illustrated in Figure You do this in the respective voice card configuration mode. The commands required to perform this initial DSP farm configuration are provided in Table DSP resources are used only if this command is configured for the particular voice card.
To delete a disabled profile, use the no form of this command. If the profile is successfully created, the user enters the DSP farm profile configuration mode. Multiple profiles can be configured for the same service.
If a profile is active, the user will not be allowed to delete the profile. The profile identifier uniquely identifies a profile. If the service type and profile identifier are not unique, a message is displayed that asks the user to choose a different profile identifier.
Within the DSP farm configuration, you need to specify the supported codecs and maximum number of sessions. This configuration directly affects the number of required DSPs, so ensure that the configuration matches the design specifications. This is done using the associate application sccp command. The DSP farm configuration mode commands are provided in Table To remove the codec, use the no form of this command. Depending on the media resource, multiple codecs can be configured. Using higher complexity codecs, such as G.
To reset to the default, use the no form of the command. For conferencing, the number specifies the number of conferences, not participants. To remove the protocol, use the no form of this command. This also requires a correct sccp group configuration to work correctly. To remove a particular server from the list, use the no form of this command. You can configure up to four Cisco Unified Communications Manager servers, a primary and up to three backups, to support DSP farm services.
To do this, use the priority option, with 1 being the highest priority and 4 being the lowest. To deselect the interface, use the no form of this command. WAN interfaces should be avoided. The port option should be used only if the default port has been changed on Cisco Unified Communications Manager. To disable the protocol, use the no form of this command.
SCCP and its related applications transcoding and conferencing become enabled only if DSP resources for these applications are configured, DSP-farm service is enabled, and the Cisco Unified Communications Manager registration process is completed. The no form of this command disables SCCP and its applications by unregistering from the active Cisco Unified Communications Manager, dropping existing connections, and freeing allocated resources.
To bind an SCCP group to a local interface, use the bind interface command. Table describes these SCCP group configuration commands. To remove a particular Cisco Unified Communications Manager group, use the no form of this command. Use this command to group Cisco Unified Communications Manager servers that are defined with the sccp ccm command. You can use the associate profile command to associate designated DSP farm profiles so that the DSP services are controlled by the Cisco Unified Communications Manager servers in the group.
The identifier-number references the Cisco Unified Communications Managers that were previously configured using the sccp ccm command. The profile option references the identifier of a DSP farm profile configured using the dspfarm profile command. The device name must match the name configured in Cisco Unified Communications Manager.
Otherwise, the profile is not registered to Cisco Unified Communications Manager. Each profile can be associated to only one Cisco Unified Communications Manager group. To unbind the selected interface, use the no form of this command. The selected interface is used for all calls that belong to the profiles associated to this Cisco Unified Communications Manager group. Interfaces are selected according to user requirements.
If only one group interface exists, configuration is not needed. Example shows two available DSPs configured for conferencing. Chapter Review Questions The answers to these review questions are in the appendix. According to the G. Identify the preferred voice quality measurement approach for VoIP networks. MOS b. PESQ c. QRT d. PSQM Which method of fax relay uses a store-and-forward approach?
Choose 3. The book gives you the information needed to implement and support data and voice integration solutions at the network-access level. Whether you are preparing for CCVP certification or simply want to gain a better understanding of VoIP fundamentals, you will benefit from the foundation information presented in this book.
Cisco Press. With 19 years of Cisco networking experience, Kevin has been a network design specialist for the Walt Disney World Resort and a network manager for Eastern Kentucky University.
Watch the author perform fundamental CVoice configuration tasks in a series of six video-on-demand labs. Books in this series provide officially developed self-study solutions to help networking professionals understand technology implementations and prepare for the Cisco Career Certifications examinations.
Download the sample pages. Effects of Voice Activity Detection on Bandwidth Configuring Conferencing and Transcoding on Voice Gateways Summarizing Examples of Voice Port Applications Centralized Automated Message Accounting Characteristics of the Default Dial Peer Private and Public Numbering Plan Integration Voice Translation Profiles Versus the dialplan-pattern Command Gatekeeper Hardware and Software Requirements Gatekeeper Transaction Message Protocol Configuring Gateways to Use H.
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