Cisco Systems H.323/SIP manual Protocol Description, Sip

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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

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

A call-control protocol that supports all existing functionality that is

(SIP)

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

An Internet protocol responsible for the transmission of real-time data,

Protocol (RTP)

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

A protocol that is used to establish connections, locate resources,

(SSP)

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

A messaging service that enables NT services on the IP-gateway server

Gateway System Integrity

to communicate directly with the Cisco Unified MeetingPlace system.

Manager (SIM)

 

 

 

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.

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

 

OL-6571-02

1-5

 

 

 

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Contents Text Part Number OL-6571-02 Corporate HeadquartersCopyright 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 Scope AudienceProduct Naming Convention New Features in This ReleaseFeature Description Naming Conventions Used in This GuideCisco Unified MeetingPlace System SIP RTP SIP Protocol DescriptionAudio 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 Step H.323 Device Description Cisco SIP IP Phone Description Additional ReferencesA P T E R CallManager Administration Choose Originator Click the Add a New Gateway linkField Description Field Description Task Tone box Server to the Cisco Unified CallManager ConfigurationChoose Start Run Select MeetingPlace IP Gateway and click Remove Configuring Cisco Unified MeetingPlace Gateway SIMChanging 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 How to Configure a Dialing Group Information About Configuring a Dialing GroupConfiguring a Dialing Group Example OL-6571-02 Troubleshooting Network Connectivity Unable to Make Calls From a Cisco IP Phone Troubleshooting Caller ConnectivityFast 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 Poor or Low-Audio Quality Troubleshooting Audio ProblemsEcho OL-6571-02 Description Value WorksheetsDescription 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.