Cisco Systems H.323/SIP manual Troubleshooting Audio Problems, Poor or Low-Audio Quality

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Chapter 4 Troubleshooting Cisco Unified MeetingPlace H.323/SIP IP Gateway Software Release 5.2.1

Troubleshooting Audio Problems

Step 5 Verify that the E.164 Address and H.323 ID fields are correct for H.323 outdials.

Step 6 Verify that the Display Name, User Name, and Session Name fields are correct for SIP outdials.

Checking Cisco Unified CallManager When Unable to Dial Out on IP Ports

Step 1 If Release 5.2.1 is installed on a gateway with multiple IP addresses, verify that Cisco Unified CallManager has an H.323 gateway configuration for each address.

Step 2 Verify that the gateway settings created for Release 5.2.1 allow dialing out.

Troubleshooting Audio Problems

See the following sections for information about troubleshooting audio problems:

•Poor or Low-Audio Quality, page 4-8

•Echo, page 4-9

Poor or Low-Audio Quality

Possible Cause—The caller is using a low-quality headset with the Cisco IP phone.

Corrective Action—Reduce the speaker volume to a volume that is comfortable but not loud enough to cause feedback from the microphone back to the other end of the call.

Corrective Action—Use a headset that is approved by Cisco Systems.

Possible Cause—Cisco IP phone audio settings need adjustment.

Corrective Action—During a meeting, on a Cisco 7960, press the blue i button twice to obtain network settings. The information that you receive provides statistics needed to optimize your network for VoIP.

Corrective Action—Lower the volume. Voice quality degrades if the volume on a Cisco IP phone is set to maximum.

Possible Cause—Network settings may need to be modified.

Corrective Action—Consider the CoS/QoS setting on your network. If the CoS setting is IP Precedence 5, you should hear considerable improvement in audio quality.

Corrective Action—Establish locations on your network. Locations enable you to regulate voice quality by limiting the amount of bandwidth that is available for calls.

For more information, refer to the Location Configuration section in the appropriate

Cisco Unified CallManager Administration Guide for your release.

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

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OL-6571-02

 

 

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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 Choose Originator Click the Add a New Gateway linkField Description 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 Settings Setting Description DefaultGeneral Settings SIP Settings Field Name Setting SIP Field Name Setting Choose Specify an IP address Assigning the Primary IP AddressField Task 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 PhoneFast Busy Signal Heard When Using a Cisco IP Phone Dead Air Heard When Using an H.323 DeviceDead Air 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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