Administration Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway, Release 5.2.1
Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Download this chapter
Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway SoftwareIntroducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software
Download the complete book
Book-length PDF: Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1Book-length PDF: Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 (PDF - 680 KB)
give_us_feedback

Table Of Contents

Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Audience

Scope

Naming Conventions Used in This Guide

New Features in This Release

Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components

Cisco Unified MeetingPlace System

Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Uses

Dual Tone Multi-Frequency Support by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Audio Quality During a Cisco Unified MeetingPlace Meeting

Endpoints That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

How PSTN and Cisco IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

How H.323 Clients and Cisco SIP IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Additional References


Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

This chapter includes the following sections:

•Audience

•Scope

•Naming Conventions Used in This Guide

•New Features in This Release

•Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components

•Additional References

Note In this guide, Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 is referred to as Release 5.2.1.

Audience

This guide is for network and telephony system administrators who are responsible for installing and configuring Release 5.2.1 for use with the Cisco Unified MeetingPlace system.

Scope

Administrator Guide for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 provides information about Release 5.2.1 that enables you to perform the following actions:

•Understand the Cisco Unified MeetingPlace system and related IP telephony components.

•Install and configure Release 5.2.1.

•Configure Cisco Unified CallManager to route IP calls to the IP-gateway server.

•Use Release 5.2.1 with IP PBX systems that are running standard H.323 or SIP call control—such as Avaya, Nortel, Alcatel, and Pingtel systems.

This guide does not provide information about configuring third-party, call-control applications. If you are using an IP PBX that runs standard H.323 or SIP call control, see the "Information About Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1" section on page 3-1 for required system settings and see your IP PBX documentation for information about how to configure those settings.

Additionally, this guide does not provide information about installing Multi Access (MA) blades or configuring the Cisco Unified MeetingPlace Audio Server system for IP; for more information about these topics, see the "Additional References" section.

Naming Conventions Used in This Guide

The following naming conventions are used in this guide:

Product
Naming Convention

Cisco Unified MeetingPlace Audio Server release and hardware upon which the release is installed

Cisco Unified MeetingPlace Audio Server system

Cisco Unified MeetingPlace Audio Server with any possible combinations of integration applications

Cisco Unified MeetingPlace system

Cisco Unified MeetingPlace Gateway System Integrity Manager

Gateway SIM

Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Release 5.2.1

Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1—the hardware upon which Release 5.2.1 is installed

IP-gateway server


New Features in This Release

Release 5.2.1 includes the following new features:

Feature
Description

Dialing Group Configuration

Dialing group configuration customizes the Cisco Unified MeetingPlace Audio Server system by presenting specific voice prompts to callers who dial in to a meeting by using a particular IP phone number.

Improved Cisco Unified MeetingPlace Gateway SIM Installation

During Release 5.2.1 installation, the Gateway SIM installs or upgrades automatically if an earlier Gateway SIM release is detected.


Information About Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Components

Supporting up to 960 IP connections, Release 5.2.1 works with the Cisco Unified MeetingPlace Audio Server system to provide meeting access to callers. The Cisco Unified MeetingPlace Audio Server system supports connections from up to sixteen IP-gateway servers; this multigateway support provides network load balancing and system redundancy.

To deploy Release 5.2.1, your network must have following system components:

•Cisco Unified MeetingPlace Audio Server system to provide conferencing functionality.

•Release 5.2.1 to perform IP call setup and tear down for the Cisco Unified MeetingPlace Audio Server system.

•Endpoints that are supported by Release 5.2.1 to connect callers to the Cisco Unified MeetingPlace Audio Server system.

•One of the following applications to route IP calls to the IP-gateway server:

–Cisco Unified CallManager

–Cisco SIP Proxy Server

–Cisco Gateway

Note If you are using an IP PBX that runs standard H.323 or SIP call control, see the "Information About Configuring Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1" section on page 3-1 for the required system settings and see your IP PBX documentation for information about how to configure these settings.

Cisco Unified MeetingPlace System

Consisting of the Cisco Unified MeetingPlace Audio Server system and a variety of integration applications, the Cisco Unified MeetingPlace system is an integrated communication and productivity tool that is deployed on a corporate network behind the firewall. With the Cisco Unified MeetingPlace system, users in different locations can collaborate in real time by sharing documents over personal computers and discussing content over telephones.

Access to the Cisco Unified MeetingPlace system is easy through end-user desktop applications, such as web browsers and instant messaging clients. The Cisco Unified MeetingPlace system also integrates with groupware clients and PSTN and IP-based telephones. Because of this access and integration, users can quickly schedule and attend Cisco Unified MeetingPlace meetings from any location by using their preferred interfaces.

For additional information about the Cisco Unified MeetingPlace system, see the Installation Planning Guide for Cisco Unified MeetingPlace 5.3 at the following URL:

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_installation_guides_list.html

Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

IP telephony uses your data network infrastructure to transmit voice packets. The underlying technology that is used by IP telephony applications is Voice over IP (VoIP), which enables different types of endpoints—IP phones, PSTN phones, and H.323 clients, for example—to communicate over your network.

The following sections provide information about VoIP concepts and how they relate to Release 5.2.1:

•Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

•Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Uses

•Dual Tone Multi-Frequency Support by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

•Audio Quality During a Cisco Unified MeetingPlace Meeting

Standards That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Release 5.2.1 supports the following networking and telephony standards:

•H.323

•SIP

•RTP

•Codec G.711 alaw and ulaw (64 kbps) and G.729a (8 kbps)

Note By default, G.729a is not enabled, and G711 codec calls are negotiated first. For more information about assigning codec preferences, see the Configuration Guide for Cisco Unified MeetingPlace Audio Server Release 5.3 at the following URL: http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_installation_and_configuration_guides_list.html

Protocols That Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1 Uses

Protocols are rules that endpoints follow for sending and receiving messages, checking errors, and compressing data. Release 5.2.1 uses the following protocols to transmit data throughout the Cisco Unified MeetingPlace system:

Protocol
Description

H.323

The protocol that is responsible for communication between Cisco Unified CallManager and Release 5.2.1. The protocol suite, which extends H.225 for call signaling and H.245 for data transfer, is used in the successful acceptance and media exchange of data.

Session Initiation Protocol (SIP)

A call-control protocol that supports all existing functionality that is available to a Cisco IP phone. Release 5.2.1 complies with RFC 3261 and RFC 3515 specifications and interoperates with the following endpoints:

•Cisco SIP Proxy Server environment

•Cisco 7960 and Cisco 7940 SIP IP phones

•Cisco IP/Videoconferencing Multipoint Control Unit (IP/VC MCU)

•Microsoft Real-Time Communications (RTC) Server for integration with Windows XP Messenger

Real-Time Transport Protocol (RTP)

An Internet protocol responsible for the transmission of real-time data, such as video and audio. Generally, RTP runs on top of User Datagram Protocol (UDP) but can also be supported by other transport protocols.

For Release 5.2.1, RTP is responsible for carrying the G.711 and G.729a encoded data. G.711 is a standard 64 kbps codec, and G.729a is an 8 kbps codec. Both codecs offer quality audio transmission over high-speed connections.

Skinny Station Protocol (SSP)

A protocol that is used to establish connections, locate resources, forward data, and handle flow control and error recovery, which enable a Cisco IP phone to notify Cisco Unified CallManager of its ability to place and receive calls.

Cisco Unified MeetingPlace Gateway System Integrity Manager (SIM)

A messaging service that enables NT services on the IP-gateway server to communicate directly with the Cisco Unified MeetingPlace system.


Dual Tone Multi-Frequency Support by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Dual Tone Multi-Frequency (DTMF) is a signaling method that allocates a specific pair of frequencies to each key on a touch-tone telephone. Various Cisco Unified MeetingPlace Audio Server system functions are invoked when callers press touch-tone keys in certain combinations. For example, the #5 key combination enables callers to mute and unmute their phones during a meeting.

PSTN phones use in-band DTMF, which embeds the tone in the audio stream. Although in-band DTMF is efficient, it cannot carry DTMF signals reliably when a voice compression codec is used.

H.323 clients can use out-of-band DTMF, which carries digitized information on a separate data channel and sends this information directly to Release 5.2.1. Because out-of-band DTMF does not require that the tone be deciphered, distortion and signal loss are minimal.

The Cisco Unified MeetingPlace system also supports RFC 2833: DTMF signals can be sent in the RTP stream by using packets designed to carry the signal characteristics. The DTMF signal is not embedded in the media and, therefore, does not suffer signal loss due to audio compression.

Release 5.2.1 handles both in-band and out-of-band DTMF.

Note Release 5.2.1 does not support out-of-band digit detection with SIP.

Audio Quality During a Cisco Unified MeetingPlace Meeting

The audio quality during a meeting depends upon the architecture of your network. Severe demands on bandwidth, overloading, and latency cause dropped packets, resulting in broken audio, congestion, and disruption of service.

In general, a switched-100 Mbps network handles VoIP traffic efficiently. To alleviate potentially disruptive service and to improve audio quality, consider implementing class of service (CoS) and quality of service (QoS).

When the server handles over 400 ports of IP calls, voice quality degradation can occur because of network congestion. CoS is a technology that helps manage network traffic by assigning a class to similar types of traffic and assigning a priority to each class. Typically in a VoIP environment, voice traffic is set to a high priority while data traffic is set to a low priority, and CoS makes a best effort to provide QoS by managing traffic based upon the assigned class and priority.

Release 5.2.1 implements IP Precedence Level 5 CoS for voice traffic. If your network is set to use this CoS, the resulting QoS maximizes audio quality during your meetings.

Note Release 5.2.1 does not support sending Layer 2 QoS or CoS; therefore, you cannot set priorities at the Layer 2 switch level.

Endpoints That are Supported by Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Release 5.2.1 integrates easily with existing networks to host Cisco Unified MeetingPlace meetings for users through the following supported endpoints:

•Cisco IP Phones

•Cisco SIP IP Phones

•H.323 clients, such as Microsoft NetMeeting

•PSTN phones through a voice gateway

How PSTN and Cisco IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

When a call is placed from a PSTN phone to a Cisco IP phone, the call is routed through a voice gateway, which is the demarcation point where the circuit-switched voice network meets the packet-switched data network. The primary responsibility of the voice gateway is to ensure that PSTN voice traffic reaches the data network and vice versa. You can use the voice gateway to forward an IP or PSTN call to its opposing network through Cisco Unified CallManager or a PBX.

When a call is placed from an Cisco IP phone, it is routed to Cisco Unified CallManager, which is responsible for setting up the call, directing the call to the called device, and sending network information— such as the IP address, UDP port number, and communication capabilities of the called device—to the Cisco IP phone. After receiving the information, the Cisco IP phone sends its digitized voice traffic directly to the called device.

The following steps describe how Cisco IP phones and PSTN phones use Release 5.2.1 to access the Cisco Unified MeetingPlace Audio Server system, as shown in Figure 1-1.

Figure 1-1 Cisco IP Phones and PSTN Phones Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software to Access the Cisco Unified MeetingPlace Audio Server System

.

Step
Cisco IP Phone Description
PSTN Phone Description

1.

On the Cisco IP phone dial pad, the caller enters a dialable number to the Cisco Unified MeetingPlace Audio Server system that will host the meeting.

By using a PSTN phone, the caller dials the number to the voice gateway.

2.

The call is immediately routed by using SSP to Cisco Unified CallManager.

The voice gateway routes the call to Cisco Unified CallManager.

3.

Cisco Unified CallManager and Release 5.2.1 communicate by using H.323. This communication process involves H.225 for call signaling and H.245 for media exchange.

Cisco Unified CallManager examines its routing table to resolve the dialed number with the IP address of the IP-gateway server.

Cisco Unified CallManager and Release 5.2.1 communicate by using H.323. This communication process involves H.225 for call signaling and H.245 for media exchange.

a. Cisco Unified CallManager and Release 5.2.1 use H.225 to determine if the Cisco Unified MeetingPlace Audio Server system can accept the call. By using Cisco Unified MeetingPlace GWSIM, Release 5.2.1 communicates directly with the Cisco Unified MeetingPlace Audio Server system to determine its availability.

b. If the Cisco Unified MeetingPlace Audio Server system is unavailable, Release 5.2.1 informs Cisco Unified CallManager, and the caller hears a fast busy signal.

c. If the call is accepted, Cisco Unified CallManager and Release 5.2.1 use H.245 to negotiate which codec will carry the voice activity. Release 5.2.1 uses G.711 or G.729a to carry the encoded speech.

d. Once codec negotiation is complete, Release 5.2.1 uses the Gateway SIM to retrieve an IP address and UDP port number from the Cisco Unified MeetingPlace Audio Server system. This IP address and UDP port number provide access to the meeting.

4.

Cisco Unified CallManager and Release 5.2.1 exchange the IP address and UDP port number of the Cisco IP phone or voice gateway and the Cisco Unified MeetingPlace Audio Server system

a. Cisco Unified CallManager sends the IP address and UDP port number of the Cisco Unified MeetingPlace Audio Server system to the Cisco IP phone or voice gateway.

b. Release 5.2.1 sends the IP address and UDP port number of the Cisco IP phone or voice gateway to the Cisco Unified MeetingPlace Audio Server system.

5.

After codec information, IP address, and UDP port number are received, the Cisco IP phone or voice gateway uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio Server system. The Cisco IP phone or voice gateway is connected to the Cisco Unified MeetingPlace Audio Server system after each device exchanges data.

How H.323 Clients and Cisco SIP IP Phones Communicate by Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

H.323 clients and Cisco SIP IP phones—which can be simultaneously deployed—communicate with Release 5.2.1 and provide another option to join a Cisco Unified MeetingPlace meeting.

The following steps describe how H.323 devices and Cisco SIP IP phones access the Cisco Unified MeetingPlace Audio Server system by using Release 5.2.1.

Figure 1-2 H.323 Device and Cisco SIP IP Phone Using Cisco Unified MeetingPlace H.323/SIP IP Gateway Software to Access the Cisco Unified MeetingPlace Audio Server System

.

Step
H.323 Device Description
Cisco SIP IP Phone Description

1.

A caller places a call from an H.323 device interface.

A caller places a call from a Cisco SIP IP phone.

2.

The H.323 device and Release 5.2.1 communicate by using H.323.

The Cisco SIP IP phone through Cisco SIP Proxy Server and Release 5.2.1 communicate by using SIP.

a. The H.323 device or Cisco SIP IP phone and Release 5.2.1 determine if the Cisco Unified MeetingPlace Audio Server system can accept the call. By using the Gateway SIM, the Release 5.2.1 communicates directly with the Cisco Unified MeetingPlace Audio Server system to determine its availability.

b. If the Cisco Unified MeetingPlace Audio Server system is unavailable, Release 5.2.1 informs the H.323 device or Cisco SIP IP phone, and depending upon system configuration, callers may hear a message informing them that the call cannot be accepted.

c. If the call is accepted, the H.323 device or Cisco SIP  IP phone and Release 5.2.1 negotiate which codec will carry the voice activity. Release 5.2.1 uses G.711 or G.729a to carry the encoded speech.

d. Once codec negotiation is complete, Release 5.2.1 retrieves an IP address and UDP port number from the Cisco Unified MeetingPlace Audio Server system by using Gateway SIM. This IP address and UDP port number provide access to the meeting.

3.

The H.323 device or Cisco SIP IP phone and Release 5.2.1 exchange IP addresses and UDP port numbers.

a. Release 5.2.1 sends the IP address and UDP port number of the Cisco Unified MeetingPlace Audio Server system to the H.323 device or Cisco SIP IP phone.

b. Release 5.2.1 sends the IP address and UDP port number of the H.323 device or Cisco SIP IP phone to the Cisco Unified MeetingPlace Audio Server system.

4.

After codec information, IP address, and UDP port number of the Cisco Unified MeetingPlace Audio Server system are received, the H.323 device or Cisco SIP IP phone uses the information to send voice traffic directly to the Cisco Unified MeetingPlace Audio Server system. The H.323 device or Cisco SIP IP phone is connected to the Cisco Unified MeetingPlace Audio Server system after each device exchanges data.

Additional References

See to the following documents for additional information:

•Administrator Guide for Cisco Unified MeetingPlace Audio Server Release 5.3

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_maintenance_guides_list.html

•Cisco Unified CallManager documentation for your release

http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/index.htm

•Cisco SIP Proxy Server documentation for your release

http://www.cisco.com/univercd/cc/td/doc/product/voice/sipproxy/index.htm

•Configuration Guide for Cisco Unified MeetingPlace Audio Server Release 5.3

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_installation_and_configuration_guides_list.html

•Guide to Cisco Unified MeetingPlace Conferencing Documentation and Support

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/products_documentation_roadmaps_list.html

•Installation Planning Guide for Cisco Unified MeetingPlace Release 5.3

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_installation_guides_list.html

•Release Notes for Cisco Unified MeetingPlace Audio Server Release 5.3

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_release_notes_list.html

•Release Notes for Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

http://www.cisco.com/en/US/products/sw/ps5664/ps5669/prod_release_notes_list.html