Cisco Systems H.323/SIP manual Audio Quality During a Cisco Unified MeetingPlace Meeting

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Chapter 1 Introducing Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

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

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

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

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Contents Corporate Headquarters Text Part Number OL-6571-02Copyright 2005-2006 Cisco Systems, Inc. All rights reserved N T E N T S Fast Busy Signal Heard When Using a Cisco IP Phone Worksheets OL-6571-02 Audience ScopeNew Features in This Release Feature DescriptionNaming Conventions Used in This Guide Product Naming ConventionCisco Unified MeetingPlace System SIP RTP Protocol Description SIPAudio Quality During a Cisco Unified MeetingPlace Meeting Step Cisco IP Phone Description Pstn Phone Description Step Cisco IP Phone Description Pstn Phone Description Step Device Description Cisco SIP IP Phone Description Additional References Step H.323 Device Description Cisco SIP IP Phone DescriptionA P T E R CallManager Administration Click the Add a New Gateway link Field DescriptionChoose Originator Field Description Task Server to the Cisco Unified CallManager Configuration Tone boxChoose Start Run Configuring Cisco Unified MeetingPlace Gateway SIM Select MeetingPlace IP Gateway and click RemoveChanging Cisco Unified MeetingPlace Gateway SIM Settings To apply the configuration settings, click OK again OL-6571-02 A P T E R Setting Description Default General SettingsSettings SIP Settings Field Name Setting SIP Field Name Setting Assigning the Primary IP Address Field TaskChoose Specify an IP address Information About Configuring a Dialing Group How to Configure a Dialing GroupConfiguring a Dialing Group Example OL-6571-02 Troubleshooting Network Connectivity Troubleshooting Caller Connectivity Unable to Make Calls From a Cisco IP PhoneDead Air Heard When Using an H.323 Device Dead Air Heard When Using a Cisco IP PhoneFast Busy Signal Heard When Using a Cisco IP Phone MP Resp. Msg=3 CPerr=0 SeqNum=0x16 Page Unable to Dial Out on IP Ports MeetingPlace IP outdial. Phone=651515 IRC=0 PSTN=46 Unit=0 Troubleshooting Audio Problems Poor or Low-Audio QualityEcho OL-6571-02 Worksheets Description ValueDescription Value MeetingPlace IP call flow Value OL-6571-02 D E IN-2 IN-3 IN-4

H.323/SIP specifications

Cisco Systems has been a leading force in the development and implementation of voice and video communication technologies, prominently featuring H.323 and SIP (Session Initiation Protocol). These protocols have become cornerstones in the realm of IP-based communication, facilitating seamless interaction across disparate devices and networks.

H.323 is a set of protocols that provides multimedia communication over packet-switched networks, such as the Internet. It supports audio, video, and data communications across IP networks, utilizing a variety of components including terminals, gateways, and multipoint control units (MCUs). One key feature of H.323 is its ability to handle both point-to-point and multipoint video conferencing. This makes it particularly suitable for enterprise applications where group communication is essential.

On the other hand, SIP is a more recent and flexible signaling protocol used for initiating, maintaining, and terminating real-time sessions that include voice, video, and messaging applications. Renowned for its simplicity and interoperability, SIP can work with various communication mediums and provides extensive capabilities for managing multimedia sessions. One of the primary advantages of SIP is its scalability, allowing it to accommodate everything from small-scale personal communications to vast corporate systems.

Both H.323 and SIP support features such as call transfer, call hold, and caller ID. H.323, however, can be more complex due to its broader array of standards and components, which might require substantial configuration. SIP, conversely, is designed to be lightweight, easily integrated with existing systems and applications, making it more user-friendly for developers.

Cisco Systems enhances these protocols through their robust telecommunication infrastructure that facilitates performance optimization, security, and unparalleled user experience. With advanced technologies like Cisco Unified Communications Manager and Cisco Webex, organizations can leverage H.323 and SIP to create cohesive communication environments. Security features such as encryption and authentication ensure that sensitive conversations remain private and secure, while Quality of Service (QoS) protocols manage bandwidth effectively to maintain consistent call quality.

Ultimately, the combination of H.323's established framework and SIP's flexibility ensures that organizations using Cisco Systems can effectively manage their communication needs, fostering collaboration and connectivity in today's fast-paced digital landscape. These protocols continue to evolve, adapting to the ever-increasing demands placed on modern communication systems.
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