Cisco AS5400XM Uživatelský manuál Strana 3

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All contents are Copyright © 1992–2006 Cisco Systems, Inc. All rights reserved. This document is Cisco Public Information. Page 3 of 22
The framework for VoIP services on the Cisco AS5400XM Universal Gateway is based on open interfaces and industry standards, and it
allows an ecosystem of partners to work together to develop innovative network services. Service providers are not locked into a single
VoIP signaling technology when they choose the Cisco AS5400XM—SIP, H.323, Media Gateway Control Protocol (MGCP), and
Trunking Gateway Control Protocol (TGCP) support are all built in, allowing service providers to enable the call-control protocol that
is the best fit for their network today, with the assurance that they can respond to evolving market requirements whenever necessary.
SIP
SIP is a peer-to-peer, multimedia signaling protocol that integrates with other Internet services, such as e-mail, voicemail, instant
messaging, multiparty conferencing, and multimedia collaboration. When used with an IP infrastructure, SIP helps enable rich
communications with numerous multivendor devices and media. SIP is the IETF standard for multimedia conferencing over IP. Defined
originally in RFC 2543 and updated with RFC 3261, SIP is an ASCII-based, application layer control protocol that can be used to
establish, maintain, and terminate calls between two or more endpoints.
Cisco has been instrumental in defining SIP standards. The company has been at the forefront of SIP technology since the first SIP IETF
RFC was published in 1999. As the IETF co-chair for multiple SIP working groups, Cisco actively contributes to SIP standards.
The SIP implementation on the Cisco AS5400XM Universal Gateway includes support for RFC 3261 as well as critical features such as
third-party call control, secure signaling using Transport Layer Security (TLS), and RFC 2833: RTP Payload for DTMF Digits, Telephony
Tones, and Telephony Signals. The Cisco AS5400XM also supports many important SIP extensions, including RFC 3262: Standard for
Reliability of Provisional Responses in SIP (PRACK), and RFC 3264: Standard for Offer/Answer Model with Session Description
Protocol (SDP).
H.323
Leading the industry through the adoption of new standards-based H.323 technology, the Cisco AS5400XM Universal Gateway supports
the feature and scalability enhancements introduced in H.323v2 and H.323v4. For example:
Media authentication and encryption using Secure Real-Time Transport Protocol (SRTP) are supported.
Multiple concurrent calls can be supported over a single H.225 call-signaling channel to reduce call-setup and call-clearing times
and increase network call capacity.
H.225 messages can be transported over TCP or User Datagram Protocol (UDP) as described in H.323 Annex E. Using UDP for
call-signaling transport effectively enables media cut-through in a single round trip.
H.225 offers the ability to report capacity statistics to the gatekeeper on a per-call basis for each DS-0, trunk group, or carrier
associated with the PSTN-side interfaces to assist in routing decisions.
H.323 operates in most VoIP backbone networks today, carrying billions of call minutes in many of the world’s largest VoIP networks.
H.323-based services continue to grow in service provider usage and profit.
Similarities Between SIP and H.323
Although SIP messages are not directly compatible with H.323, both protocols can coexist in the same packet telephony network because
the Cisco AS5400XM Universal Gateway can process individual SIP and H.323 calls simultaneously, allowing service providers to
integrate complementary H.323 and SIP services in the same network.
Both H323 and SIP were designed to address session control and signaling functions in a distributed call-control architecture.
Both are especially well-suited for communication with intelligent network endpoints.
Both protocols are essential for solutions where an intelligent media gateway is used for PSTN termination.
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