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Cisco Unified Communications SRND Based on Cisco Unified Communications Manager 6.x
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Preface
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Introduction
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Unified Communications Deployment Models
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Network Infrastructure
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Gateways
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Cisco Unified CM Trunks
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Media Resources
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Music on Hold
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Call Processing
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Call Admission Control
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Dial Plan
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Emergency Services
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Third-Party Voicemail Design
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Cisco Unity
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Cisco Unity Express
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Cisco Unified MeetingPlace Integration
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Cisco Unified MeetingPlace Express
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IP Video Telephony
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LDAP Directory Integration
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IP Telephony Migration Options
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Voice Security
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Unified Communications Endpoints
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Device Mobility
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Cisco Unified Presence
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Cisco Unified CM Applications
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Cisco Mobility Applications
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Network Management
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Cisco Unified Communications Manager Business Edition
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Recommended Hardware and Software Combinations
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Glossary
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Index
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Table Of Contents
Traffic Patterns and Gateway Sizing
Normal Business Traffic Profile
Contact Center Traffic Profile
Gateway Sizing for Contact Center Traffic
Voice Activity Detection (VAD)
Cisco Access Digital Trunk Gateways
Gateway Protocols and Core Feature Requirements
Site-Specific Gateway Requirements
Gateway Support for Fax Passthrough and Fax Relay
Gateway Support for Modem Passthrough and Modem Relay
Supported Platforms and Features
Gateway Configuration Examples
Cisco IOS Gateway Configuration for Modem Passthrough
Cisco VG248 Configuration for Modem Passthrough
Unified CM Configuration for the Cisco 6608 and 6624 Non-IOS Gateways
Clock Sourcing for Fax and Modem Passthrough
Named Service Event (NSE) T.38 Fax Relay
Routing Inbound Calls from the PSTN
Routing Outbound Calls to the PSTN
Automated Alternate Routing (AAR)
ISDN B-Channel Binding, Rollover, and Busy Out
Configuring the Gateways in Unified CM
Voice Gateways Bearer Capabilities
Gateways
Gateways provide a number of methods for connecting an IP telephony network to the Public Switched Telephone Network (PSTN), a legacy PBX, or key systems. Gateways range from specialized, entry-level and stand-alone voice gateways to high-end, feature-rich integrated router and Cisco Catalyst gateways.
This chapter explains important factors to consider when selecting a Cisco gateway to provide the appropriate protocol and feature support for your IP Telephony network. The main topics discussed in this chapter include:
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Traffic Patterns and Gateway Sizing
Traffic Patterns and Gateway Sizing
This section presents a high-level discussion of the differences between various traffic models or patterns and how they can affect voice gateway selection. The emphasis is on traffic patterns and gateway sizing for traffic-intensive deployments.
Definitions and Terminology
This section uses the following terms and definitions:
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The number of calls that are all active in the system at the same time.
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The maximum number of simultaneous calls in active (talk) state that the system can handle. The number of calls expected to be active simultaneously during the busy hour of the day should not exceed this number.
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The call arrival rate, described as the number of calls that arrive (that is, new call setup attempts) in one second. Call arrival rates are also often quoted in calls per hour, but this metric is looser in the sense that 100 calls arriving in the last five seconds of an hour provides an average call arrival rate of 100 calls per hour (which is an extremely low rate for a communications system), while it also provides an arrival rate of 20 calls per second (which is a high rate). Sustaining 20 calls per second for an entire hour would result in 72,000 calls per hour. Therefore calls-per-hour is not a very useful metric for ascertaining a system's ability to handle bursty call arrival traffic patterns.
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The number of calls attempted during the busiest hour of the day (the peak hour). This is the same as the calls-per-second rating for the busiest hour of the day, but it is expressed over a period of an hour rather than a second. For example, 10 cps would be equal to 3600 calls per hour. There is also a metric for Busy Hour Call Completions (BHCC), which can be lower than the BHCA (call attempts) under the assumption that not all calls are successful (as when a blocking factor exists). This chapter assumes 100% call completions, so that BHCA = BHCC.
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Steady arrival means the call attempts are spaced more or less equally over a period of time. For example, 60 calls per hour at a steady arrival rate would present one call attempt roughly every minute (or approximately 0.02 cps). With bursty arrival, the calls arriving over a given period of time (such as an hour) are not spaced equally but are clumped together in one or more spikes. In the worst case, an arrival rate of 60 calls per hour could offer all 60 calls in a single second of the hour, thus averaging 0 cps for most of the hour with a peak of 60 cps for that one second. This kind of traffic is extremely stressful to communications systems.
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This is the period of "talk time" on a voice call; that is, the period of time between call setup and call teardown when there is an open speech path between the two parties. A hold time of 3 minutes (180 seconds) is an industry average used for traffic engineering of voice systems. The shorter the hold time on the average call, the greater the percentage of system CPU time spent on setting up and tearing down calls compared to the CPU time spent on maintaining the speech path.
PSTN Traffic Patterns
Traffic, when used in the context of voice communication systems, refers to the volume of calls being sent and/or received. Of particular importance is the traffic carried by external circuits such as the public switched telephone network (PSTN). Traffic is measured in Erlangs, and an Erlang is defined as one call lasting for one hour. This section does not go into any further detail on Erlangs other than to say that there are mathematical tables (Erlang-B and Erlang-C) that are used to calculate how many circuits are required for a given amount of offered traffic.
The amount of traffic received and generated by your business determines the size of the external circuits required. However; many customers typically continue to use the same number of circuits for their IP-based communications system as they previously used for a TDM-based system. While this sizing method might work if no issues are encountered, the process of ongoing system traffic analysis should be part of any routine maintenance practices. Traffic analysis can show that the system is over-provisioned for the current levels of traffic (and, therefore, the customer is paying for circuits that are not needed) or, conversely, that the system is under-provisioned and may be suffering from occasional blocked and/or lost calls, in which case increasing the number of circuits will remedy the situation.
Normal Business Traffic Profile
Most customers have a normal traffic profile, which means that they typically have two busy hours per day, one occurring during the morning from 10:00 to 11:00 and the other in the afternoon from 14:00 to 15:00. These busy-hour patterns can often be attributed to such things as employees starting the work day or returning from a lunch break. The calls themselves tend to have longer hold times and they tend to arrive and leave in a steady manner. A generally accepted industry average holding time to use for traffic calculations is 3 minutes.
Assuming that the communications system is engineered with the busy-hour traffic in mind, no issues should arise. Engineering a system below these levels will result in blocked and/or lost calls, which can have a detrimental effect on business.
Contact Center Traffic Profile
Contact centers present somewhat different patterns of traffic in that these systems typically handle large volumes of calls for the given number of agents or interactive voice response (IVR) systems available to service them. Contact centers want to get the most out of their resources, therefore their agents, trunks, and IVR systems are kept busy all the while they are in operation, which usually is 24 hours a day. Call queuing is typical (when incoming call traffic exceeds agent capacity, calls wait in queue for the next available agent), and the agents are usually dedicated during their work shifts to taking contact center calls.
Call holding times in contact centers are often of a shorter average duration than normal business calls. Contributing to the shorter average call holding time is the fact that many calls interact only with the IVR system and never need to speak to a human agent (also termed self-service calls). A representative holding time for self-service calls is about 30 seconds, while a call that talks to an agent has an average holding time of 3 minutes (the same as normal business traffic), making the overall average holding time in the contact center shorter than for normal business traffic.
The goal of contact centers to optimize resource use (including IVR ports, PSTN trunks, and human agents), combined with the fact that contact centers are systems dedicated to taking telephone calls, also presents the system with higher call arrival rates than in a typical business environment. These call arrival rates can also peak at different times of day and for different reasons (not the usual busy hour) than normal business traffic. For example, when a television advertisement runs for a particular holiday package with a 1-800 number, the call arrival rate for the system where those calls are received will experience a peak of traffic for about 15 minutes after the ad airs. This call arrival rate can exceed the average call arrival rate of the contact center by an order of magnitude.
Gateway Sizing for Contact Center Traffic
Short call durations as well as bursty call arrival rates impact the PSTN gateway's ability to process the traffic. Under these circumstances the gateway needs more resources to process all calls in a timely manner, as compared to gateways that receive calls of longer duration that are presented more uniformly over time. Because gateways have varying capabilities to deal with these traffic patterns, careful consideration should be given to selecting the appropriate gateway for the environment in which it will operate. Some gateways support more T1/E1 ports than others, and some are more able than others to deal with multiple calls arriving at the same time.
For a traffic pattern with multiple calls arriving in close proximity to each other (that is, high or bursty call arrival rates), a gateway with a suitable rating of calls per second (cps) is the best fit. Under these conditions, using calls with 15-second hold times, the Cisco AS5400XM Universal Gateway can maintain 20 cps (with 310 calls active at once), the Cisco 3845 Integrated Services Router can maintain 17 cps (with 255 calls active at once), and the Cisco Catalyst 6500 Communication Media Module can maintain 7 cps (with 130 calls active at once). The performance of the Cisco AS5350XM Universal Gateway is identical to that of the AS5400XM in terms of calls per second.
For traffic patterns with a steady arrival rate, the maximum number of active calls that a gateway can handle is generally the more important consideration. Under these conditions, using calls with 180-second hold times, the Cisco AS5400XM Universal Gateway can maintain 630 simultaneously active calls (with a call arrival rate of up to 3.5 cps), the Cisco 3845 Integrated Services Router can maintain 504 simultaneously active calls (with a call arrival rate of up to 3 cps), and the Cisco Catalyst 6500 Communication Media Module can maintain 240 simultaneously active calls (with a call arrival rate of up to 1.3 cps).
These numbers assume that all of the following conditions apply:
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Voice Activity Detection (VAD)
VAD is a digital signal processing feature that suppresses the creation of most of the IP packets during times when the speech path in a particular direction of the call is perceived as being silent. Typically only one party on a call speaks at a time, so that packets need flow in only one direction, and packets in the reverse (or silent) direction need not be sent except as an occasional keepalive measure. VAD can therefore provide significant savings in the number of IP packets sent for a VoIP call, and thereby save considerable CPU cycles on the gateway platform. While the actual packet savings that VAD can provide varies with the call flow, the application, and the nature of speaker interactions, it tends to use 10% to 30% fewer packets than would be sent for a call made using a VAD-off configuration.
VAD is most often turned off in endpoints and voice gateways deployed in Unified CM networks; VAD is most often turned on in voice gateways in other types of network deployments.
Codec
Both G.711 and G.729A use as their default configuration a 20 ms sampling time, which results in a 50 packets per second (pps) VoIP call in each direction. While a G.711 IP packet (200 bytes) is larger than a G.729A packet (60 bytes), this difference has not proven to have any significant effect on voice gateway CPU performance. Both G.711 and G.729 packets qualify as "small" IP packets to the router, therefore the packet rate is the salient codec parameter affecting CPU performance.
Performance Overload
Cisco IOS is designed to have some amount of CPU left over during peak processing, to handle interrupt-level events. The performance figures in this section are designed with the processor running at an average load of approximately 75%. If the load on a given Cisco IOS gateway continually exceeds this threshold, the following will result:
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Cisco IOS Gateways are designed to handle a short burst of calls, but continual overloading of the recommended call rate (calls per second) is not supported.
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Performance Tuning
The CPU utilization of a Cisco IOS Voice gateway is affected by every process that is enabled in a chassis. Some of the lowest level processes such as IP routing and memory defragmentation will occur even when there is no live traffic on the chassis.
Lowering the CPU utilization can help to increase the performance of a Cisco IOS Voice Gateway by ensuring that there are enough available CPU resources to process the real-time voice packets and the call setup instructions. Some of the techniques for decreasing CPU utilization are described in Table 4-1.
Additional Information
For more information on Cisco Voice Gateway capabilities and call center traffic analysis, refer to the following sources:
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http://www.cisco.com/application/pdf/en/us/guest/products/ps259/c1650/cdccont_0900aecd8057f2e0.pdf
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http://www.cisco.com/en/US/products/hw/univgate/ps505/products_data_sheet0900aecd802efc92.html
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http://www.cisco.com/en/US/products/hw/univgate/ps505/prod_brochure0900aecd802f6ece.html
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http://www.erlang.com/calculator/
Understanding Cisco Gateways
Cisco access gateways enable Cisco Unified Communications Manager (Unified CM) to communicate with non-IP telecommunications devices. There are two types of Cisco access gateways, analog and digital.
Cisco Access Analog Gateways
There are two categories of Cisco access analog gateways, trunk gateways and station gateways.
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Analog station gateways connect Unified CM to Plain Old Telephone Service (POTS) analog telephones, interactive voice response (IVR) systems, fax machines, and voice mail systems. Station gateways provide Foreign Exchange Station (FXS) ports.
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Analog trunk gateways connect Unified CM to PSTN central office (CO) or PBX trunks. Trunk gateways provide Foreign Exchange Office (FXO) ports for access to the PSTN, PBXs, or key systems, and E&M (recEive and transMit, or ear and mouth) ports for analog trunk connection to a legacy PBX. Whenever possible, use digital gateways to minimize any answer and disconnect supervision issues. Analog Direct Inward Dialing (DID) and Centralized Automatic Message Accounting (CAMA) are also available for PSTN connectivity.
Cisco Access Digital Trunk Gateways
A Cisco access digital trunk gateway connects Unified CM to the PSTN or to a PBX via digital trunks such as Primary Rate Interface (PRI), Basic Rate Interface (BRI), or T1 Channel Associated Signaling (CAS). Digital T1 PRI trunks may also be use to connect to certain legacy voice mail systems.
Gateway Selection
When selecting an IP telephony gateway, consider the following factors:
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Core Feature Requirements
Gateways used in IP telephony applications must meet the following core feature requirements:
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DTMF relay capability, specifically out-of-band DTMF, separates DTMF digits from the voice stream and sends them as signaling indications through the gateway protocol (H.323, SCCP, MGCP, or SIP) signaling channel instead of as part of the voice stream or bearer traffic. Out-of-band DTMF is required when using a low bit-rate codec for voice compression because the potential exists for DTMF signal loss or distortion.
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Supplementary services are typically basic telephony functions such as hold, transfer, and conferencing.
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Fax over IP enables interoperability of traditional analog fax machines with IP telephony networks. The fax image is converted from an analog signal and is carried as digital data over the packet network. For more information, see Fax and Modem Support
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Cisco Unified Communications is based on a distributed model for high availability. Unified CM clusters provide for Unified CM redundancy. The gateways must support the ability to "re-home" to a secondary Unified CM in the event that a primary Unified CM fails. Redundancy differs from call survivability in the event of a Unified CM or network failure.
Refer to the gateway product documentation to verify that any IP Telephony gateway you select for an enterprise deployment can support the preceding core requirements. Additionally, every IP Telephony implementation has its own site-specific feature requirements, such as analog or digital access, DID, and capacity requirements (see Site-Specific Gateway Requirements).
Gateway Protocols
Cisco Unified CM (Release 3.1 and later) supports the following gateway protocols:
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Cisco Unified CM Release 4.0 and later supports Session Initiation Protocol (SIP) on the trunk side. The SIP trunk implementation has been enhanced in Cisco Unified CM Release 5.0 to support more features.
Cisco Unified IP Phones use Skinny Client Control Protocol (SCCP), which is a lighter-weight protocol. SCCP and MGCP use a master/slave model, while SIP and H.323 use a peer-to-peer model.
Protocol selection depends on site-specific requirements and the installed base of equipment. For example, most remote branch locations have Cisco 2600XM, 2800, 3700, or 3800 Series routers installed. These routers support SIP, H.323, and MGCP 0.1 with Cisco IOS Release 12.2.11(T) and Cisco Unified CM Release 3.1 or later. For gateway configuration, MGCP might be preferred over H.323 or SIP due to simpler configuration. On the other hand, H.323 or SIP might be preferred over MGCP because of the robustness of the interfaces supported.
Simplified Message Desk Interface (SMDI) is a standard for integrating voice mail systems to PBXs or Centrex systems. Connecting to a voice mail system via SMDI and using either analog FXS or digital T1 PRI would require either SCCP or MGCP protocol because H.323 or SIP devices do not identify the specific line being used from a group of ports. Use of H.323 or SIP gateways for this purpose means the Cisco Message Interface cannot correctly correlate the SMDI information with the actual port or channel being used for an incoming call.
In addition, the Unified CM deployment model being used can influence gateway protocol selection. (Refer to the chapter on Unified Communications Deployment Models, page 2-1.)
Table 4-2 shows which gateways support a given protocol. Each of these protocols follows a slightly different methodology to provide support for the core gateway requirements. Gateway Protocols and Core Feature Requirements, describes how each protocol provides these feature requirements.
Table 4-2 Supported Gateway Protocols and Cisco Unified Communications Gateways
Cisco 3800
Yes, beginning with Cisco IOS Release 12.3.11T
Supported with:
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Yes, beginning with Cisco IOS Release 12.3.11T
Yes for DSP resources, beginning with Cisco IOS Release 12.3.11T.
For FXS, use Cisco IOS Release 12.4.9.T.
Yes, SIP trunk
Cisco 2800
Yes, beginning with Cisco IOS Release 12.3.8T4
Supported with:
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Yes, beginning with Cisco IOS Release 12.3.8T4
Yes for DSP resources, beginning with Cisco IOS Release 12.3.11T.
For FXS, use Cisco IOS Release 12.4.9.T.
Yes, SIP trunk
Cisco 3700
Yes
Supported with:
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Yes
DSP farm in Cisco IOS Release 12.2.13T
Yes, SIP trunk
Communication Media Module (CMM)
Yes
Supported with:
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Yes
No
Yes
Catalyst 6000
WS-X6608-x1 Gateway Module and
FXS Module WS-X66241Yes
Supported with:
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No
No
No
VG224
Yes, FXS only.
Also supports conferencing and transcoding for VG224 beginning with Cisco IOS Release 12.3(T).
Yes, FXS only
Yes, beginning with Cisco IOS Release 12.4(2)T
Yes, SIP trunk
VG248
No
No
Yes2
No
Cisco ATA 188
Yes, FXS only
Yes, FXS only
Yes, FXS only
Yes, third-party SIP phone
Cisco AS5350XM
Cisco AS5400XM
No
Yes
No
Yes, SIP trunk
Cisco AS58501
No
Yes
No
Yes, SIP trunk
Cisco 53001
No
Yes
No
Yes, SIP trunk
Cisco 3640 and 36601
Yes
Supported with:
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Yes
DSP farm in Cisco IOS Release 12.2.13T
Yes, SIP trunk
Yes
Supported with:
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Yes
DSP farm in Cisco IOS Release 12.2.13T
Yes, SIP trunk
Cisco 1751 and 17601
Yes
Yes
Yes, conferencing and transcoding
Yes, SIP trunk
VG2001
Yes
Supported with:
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Yes
Yes (DSP farm)
No
Cisco 7200
No
Yes
No
Yes, SIP trunk
Catalyst 4000 WS-X4604-GWY Gateway Module1
Yes
Yes
No
No
Cisco ICS7750-MRP1
No
Yes
No
No
Cisco ICS7750-ASI1
No
Yes
No
No
DE-30+, DT-24+1
Yes
No
No
No
Cisco 827-V41
No
Yes, supported for FXS
No
No
1 These models have reached End of Sale (EOS).
2 The VG248 is not a true gateway in that it uses Skinny Client Control Protocol (SCCP) instead of H.323, MGCP, or SIP.
3 For IP Telephony applications, use Cisco 2800 Series Routers. For memory considerations for the Cisco 2600 routers, see the Product Bulletin at /en/US/products/hw/routers/ps259/prod_bulletin09186a0080088755.html
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Gateway Protocols and Core Feature Requirements
This section describes how each protocol (SCCP, H.323, MGCP, and SIP) supports the following gateway feature requirements:
DTMF Relay
Dual-Tone Multifrequency (DTMF) is a signaling method that uses specific pairs of frequencies within the voice band for signals. A 64 kbps pulse code modulation (PCM) voice channel can carry these signals without difficulty. However, when using a low bite-rate codec for voice compression, the potential exists for DTMF signal loss or distortion. An out-of-band signaling method for carrying DTMF tones across a Voice over IP (VoIP) infrastructure provides an elegant solution for these codec-induced symptoms.
SCCP Gateways
The SCCP gateways, such as the Cisco VG248, carry DTMF signals out-of-band using Transmission Control Protocol (TCP) port 2002. Out-of-band DTMF is the default gateway configuration mode for the VG248.
H.323 Gateways
The H.323 gateways, such as the Cisco 3800 series products, can communicate with Unified CM using the enhanced H.245 capability for exchanging DTMF signals out-of-band. The following is an example out-of-band DTMF configuration on a Cisco IOS gateway:
dial-peer voice 100 voipdestination-pattern 555....session target ipv4:10.1.1.1CODEC g729ar8dtmf-relay h245-alphanumericpreference 0MGCP Gateway
The Cisco IOS-based VG224, 2600XM, 2800, 3700, and 3800 platforms use MGCP for Unified CM communication. Within the MGCP protocol is the concept of packages. The MGCP gateway loads the DTMF package upon start-up. The MGCP gateway sends symbols over the control channel to represent any DTMF tones it receives. Unified CM then interprets these signals and passes on the DTMF signals, out-of-band, to the signaling endpoint. The global configuration command for DTMF relay is:
mgcp dtmf-relay CODEC all mode out-of-bandYou must enter additional configuration parameters in the Unified CM MGCP gateway configuration interface.
The Catalyst 6000, DE-30+, and DT-24+ all support MGCP with Unified CM Release 3.1 and later. DTMF relay is enabled by default and does not need additional configuration.
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SIP Gateway
The Cisco IOS-based VG224, 2600XM, 2800, 3700, 3800 platforms can use SIP for Unified CM communication. They support various methods for DTMF, but only the following methods can be used to communicate with Unified CM:
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The following example shows a configuration for NTE:
dial-peer voice 100 voipdestination-pattern 555....session target ipv4:10.1.1.1session protocol sipv2dtmf-relay rtp-nteThe following example shows a configuration for UN:
dial-peer voice 100 voipdestination-pattern 555....session target ipv4:10.1.1.1session protocol sipv2dtmf-relay sip-notifyFor more details on DTMF method selection, see the chapter on Media Resources, page 6-1.
Supplementary Services
Supplementary services provide user functions such as hold, transfer, and conferencing. These are considered fundamental requirements of any voice installation. Each gateway evaluated for use in an IP telephony network should provide support for supplementary services natively, without the use of a software media termination point (MTP).
SCCP Gateways
The Cisco VG224, VG248, and ATA 188 gateways provide full supplementary service support. The Cisco 2800 and 3800 Series gateways also support FXS SCCP ports with Cisco IOS Release 12.4.9T. The SCCP gateways use the Gateway-to-Unified CM signaling channel and SCCP to exchange call control parameters.
H.323 Gateways
H.323v2 implements Open/Close LogicalChannel and the emptyCapabilitySet features. The use of H.323v2 by H.323 gateways, beginning in Cisco IOS Release 12.0(7)T and Cisco Unified CM Release 3.0 and later, eliminates the requirement for an MTP to provide supplementary services. With Unified CM Release 3.1 and later, a transcoder is allocated dynamically only if required during a call to provide access to G.711-only devices while still maintaining a G.729 stream across the WAN. Full support for H.323v2 is available in Cisco IOS Release 12.1.1T.
Once an H.323v2 call is set up between a Cisco IOS gateway and an IP phone, using the Unified CM as an H.323 proxy, the IP phone can request to modify the bearer connection. Because the Real-Time Transport Protocol (RTP) stream is directly connected to the IP phone from the Cisco IOS gateway, a supported voice codec can be negotiated.
Figure 4-1 and the following steps illustrate a call transfer between two IP phones:
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Figure 4-1 H.323 Gateway Supplementary Service Support
MGCP Gateway
The MGCP gateways provide full support for the hold, transfer, and conference features through the MGCP protocol. Because MGCP is a master/slave protocol with Unified CM controlling all session intelligence, Unified CM can easily manipulate MGCP gateway voice connections. If an IP telephony endpoint (for example, an IP phone) needs to modify the session (for example, transfer the call to another endpoint), the endpoint would notify Unified CM using SCCP. Unified CM then informs the MGCP gateway, using the MGCP User Datagram Protocol (UDP) control connection, to terminate the current RTP stream associated with the Session ID and to start a new media session with the new endpoint information. Figure 4-2 illustrates the protocols exchanged between the MGCP gateway, endpoints, and Unified CM.
Figure 4-2 MGCP Gateway Supplementary Service Support
SIP Gateway
The Unified CM SIP trunk interface to Cisco IOS SIP gateways supports supplementary services such as hold, blind transfer, and attended transfer. The support for supplementary services is achieved via SIP methods such as INVITE and REFER. For more details, refer to the following documentation:
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Unified CM Redundancy
An integral piece of the IP telephony architecture is the provisioning of low-cost, distributed PC-based systems to replace expensive and proprietary legacy PBX systems. This distributed design lends itself to the robust fault tolerant architecture of clustered Unified CMs. Even in its most simplistic form (a two-system cluster), a secondary Unified CM should be able to pick up control of all gateways initially managed by the primary Unified CM.
SCCP Gateways
Upon boot-up, the Cisco VG224, VG248, and ATA 188 gateways are provisioned with Unified CM server information. When these gateways initialize, a list of Unified CMs is downloaded to the gateways. This list is prioritized into a primary Unified CM and secondary Unified CM. In the event that the primary Unified CM becomes unreachable, the gateway registers with the secondary Unified CM.
H.323 VoIP Call Preservation Enhancements for WAN Link Failures
H.323 VoIP call preservation enhancements for WAN link failures sustain connectivity for H.323 topologies where signaling is handled by an entity that is different from the other endpoint, such as a gatekeeper that provides routed signaling or a call agent (such as the Cisco BTS 10200 Softswitch, Cisco PGW2200 Softswitch, or Cisco Unified CM) that brokers signaling between the two connected parties. Call preservation is useful when a gateway and the other endpoint (typically a Cisco Unified IP Phone) are located at the same site but the call agent is remote and therefore more likely to experience connectivity failures.
H.323 call preservation covers the following types of failures and connections.
Failure Types:
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Connection Types:
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Note that, after the media is preserved, the call is torn down later when either one of the parties hangs up or media inactivity is detected. In cases where there is a machine-generated media stream, such as music streaming from a media server, the media inactivity detection will not work and the call might hang. Cisco Unified CM addresses such conditions by indicating to the gateway that such calls should not be preserved, but third-party devices or the Cisco Unified Border Element would not do this.
Flapping is defined for this feature as the repeated and temporary loss of IP connectivity, which can be caused by WAN or LAN failures. H.323 VoIP calls between a Cisco IOS gateway and Cisco Unified CM may be torn down when flapping occurs. When Unified CM detects that the TCP connection is lost, it clears the call and closes the TCP sockets used for the call by sending a TCP FIN, without sending an H.225.0 Release Complete or H.245 End Session message. This is called quiet clearing. The TCP FIN sent from Unified CM could reach the gateway if the network comes up for a short duration, and the gateway will tear down the call. Even if the TCP FIN does not reach the gateway, the TCP keepalives sent from the gateway could reach Unified CM when the network comes up. Unified CM will send TCP RST messages in response to the keepalives because it has already closed the TCP connection. The gateway will tear down H.323 calls if it receives the RST message.
Configuration of H.323 VoIP call preservation enhancements for WAN link failures involves configuring the call preserve command. If you are using Cisco Unified Communications Manager, you must enable the Allow Peer to Preserve H.323 Calls parameter from the Service Parameters window.
The call preserve command causes the gateway to ignore socket closure or socket errors on H.225.0 or H.245 connections for active calls, thus allowing the socket to be closed without tearing down calls using those connections.
Example of H.323 VoIP Call Preservation for All Calls
The following configuration example enables H.323 VoIP call preservation for all calls:
voice service voiph323call preserveMGCP Gateway
MGCP gateways also have the ability to fail over to a secondary Unified CM in the event of communication loss with the primary Unified CM. When the failover occurs, active calls are preserved.
Within the MGCP gateway configuration file, the primary Unified CM is identified using the call-agent <hostname> command, and a list of secondary Unified CM is added using the ccm-manager redundant-host command. Keepalives with the primary Unified CM are through the MGCP application-level keepalive mechanism, whereby the MGCP gateway sends an empty MGCP notify (NTFY) message to Unified CM and waits for an acknowledgement. Keepalive with the backup Unified CMs is through the TCP keepalive mechanism.
If the primary Unified CM becomes available at a later time, the MGCP gateway can "re-home," or switch back to the original Unified CM. This re-homing can occur either immediately, after a configurable amount of time, or only when all connected sessions have been released. This is enabled through the following global configuration commands:
ccm-manager redundant-host <hostname1 | ipaddress1 > <hostname2 | ipaddress2>[no] call-manager redundancy switchback [immediate|graceful|delay <delay_time>]SIP Gateway
Redundancy with Cisco IOS SIP gateways can be achieved similarly to H.323. If the SIP gateway cannot establish a connection to the primary Unified CM, it tries a second Unified CM defined under another dial-peer statement with a higher preference.
By default the Cisco IOS SIP gateway transmits the SIP INVITE request 6 times to the Unified CM IP address configured under the dial-peer. If the SIP gateway does not receive a response from that Unified CM, it will try to contact the Unified CM configured under the other dial-peer with a higher preference.
Cisco IOS SIP gateways wait for the SIP 100 response to an INVITE for a period of 500 ms. By default, it can take up to 3 seconds for the Cisco IOS SIP gateway to reach the backup Unified CM. You can change the SIP INVITE retry attempts under the sip-ua configuration by using the command retry invite <number>. You can also change the period that the Cisco IOS SIP gateway waits for a SIP 100 response to a SIP INVITE request by using the command timers trying <time> under the sip-ua configuration.
One other way to speed up the failover to the backup Unified CM is to configure the command monitor probe icmp-ping under the dial-peer statement. If Unified CM does not respond to an Internet Control Message Protocol (ICMP) echo message (ping), the dial-peer will be shut down. This command is useful only when the Unified CM is not reachable. ICMP echo messages are sent every 10 seconds.
The following commands enable you to configure Unified CM redundancy on a Cisco IOS SIP gateway:
sip-uaretry invite <number>timers trying <time>dial-peer voice 101 voipdestination-pattern 2...session target ipv4:10.1.1.101preference 0monitor probe icmp-pingsession protocol sipv2dial-peer voice 102 voipdestination-pattern 2...session target ipv4:10.1.1.102preference 1monitor probe icmp-pingsession protocol sipv2Site-Specific Gateway Requirements
Each IP Telephony implementation has its own site-specific requirements. The following questions can help you with IP Telephony gateway selection:
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Note
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Cisco Unified Communications gateways are able to inter-operate with most major PBX vendors, and they are EIA/TIA-464B compliant.
The site-specific and core gateway requirements are a good start to help narrow the possible choices. Once you have defined the required features, you can make a gateway selection for each of the pertinent configurations, whether they are single-site enterprise deployments of various sizes and complexities or multisite enterprise deployments.
The following tables summarize the features and interface types supported by the various Cisco gateway models.
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http://www.cisco.com/en/US/products/sw/voicesw/ps556/prod_software_versions_comparison.htmlCisco Analog Gateways
Table 4-3 lists supported interface types for Cisco analog gateways using H.323 or Session Initiation Protocol (SIP), and Table 4-4 lists supported interface types for Cisco analog gateways using Media Gateway Control Protocol (MGCP).
Table 4-3 Supported Analog H.323 and SIP Features
3800 Series
Yes
Yes
Yes
Yes
Yes
Yes
2800 Series
Yes
Yes
Yes
Yes
Yes
Yes
3700 Series1
Yes
Yes
Yes
Yes
Yes
Yes
Communication Media Module (CMM) 24FXS
Yes
N/A
N/A
N/A
N/A
N/A
CMM-6T1/E1
N/A
N/A
N/A
N/A
N/A
N/A
6608 and 66241
N/A
N/A
N/A
N/A
N/A
N/A
VG224
Yes
N/A
N/A
N/A
N/A
N/A
VG248
No
No
No
No
No
No
Analog Telephone Adapter (ATA)1
Yes
No
No
No
No
No
3600 Series1
Yes
Yes
Yes
Yes
Yes
12.2.11T
2600 Series1
Yes
Yes
Yes
Yes
Yes
12.2.11T
1751 and 17601
Yes
Yes
Yes
Yes
Yes
Yes
VG2001
Yes
Yes
Yes
No
Yes
No
7x00 family
N/A
N/A
N/A
N/A
N/A
N/A
ICS 77501
Yes
Yes
Yes
Yes
Yes
No
Catalyst 4000 Access Gateway Module (AGM)
Yes
Yes
No
No
No
No
827-4V1
Yes
No
No
No
No
No
1 These models have reached End of Sale (EOS).
Table 4-4 Supported Analog MGCP Features
3800 Series
Yes
Yes
No
Yes
No
No
2800 Series
Yes
Yes
No
Yes
No
No
3700 Series1
Yes
Yes
No
Yes
No
No
Communication Media Module (CMM) 24FXS
Yes
N/A
N/A
N/A
N/A
N/A
CMM-6T1/E1
N/A
N/A
N/A
N/A
N/A
N/A
6608 and 66241
Yes
No
No
No
No
No
VG224
Yes
No
No
No
No
No
VG248
No
No
No
No
No
No
Analog Telephone Adapter (ATA)1
Yes
N/A
N/A
N/A
N/A
N/A
3600 Series1
Yes
Yes
No
Yes
No
No
2600 Series1
Yes
Yes
No
Yes
No
No
1751 and 17601
Yes
Yes
No
Yes
No
No
VG2001
Yes
Yes
No
Yes
No
No
7x00 family
N/A
N/A
N/A
N/A
N/A
N/A
ICS 77501
Yes
Yes
No
No
No
No
Catalyst 4000 Access Gateway Module (AGM)1
Yes
Yes
No
No
No
No
827-4V1
No
No
N/A
N/A
N/A
N/A
1 These models have reached End of Sale (EOS).
Cisco Digital Gateways
Table 4-5 through Table 4-8 list supported interface types for Cisco digital gateways using H.323 or Session Initiation Protocol (SIP). Table 4-9 lists supported interface types for Cisco digital gateways using Media Gateway Control Protocol (MGCP).
