The Cisco AS5300 Series includes the Cisco AS5350 and AS5350XM Universal Gateways. The AS5350XM provides double the performance of the Cisco AS5350 allowing current applications to run faster and at a lower CPU utilization.
Cisco AS5300 Series Universal Gateways
产品简介：
- 产品基本信息
- 产品特点与优点
The Cisco AS5300 Series Universal Gateways are the only universal port-ready, one rack-unit (RU) dual T1/E1 gateways that provide carrier-class reliability in a modular design. Service provider data and voice applications are also supported, including:
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Features And Solutions
Packet Telephony
With more than 24 million voice gateway ports shipped, Cisco is a proven leader in product innovation for packet telephony services. Cisco AS5000 Universal Gateways are a primary component in many Cisco end-to-end voice solutions:
• Cisco Voice Infrastructure and Applications (VIA)
• Business voice services
• Residential voice-over-broadband (VoBB) PSTN termination services
• Contact center in combination with Cisco Unified Customer Voice Portal (CVP)
The framework for VoIP services on the Cisco AS5350XM 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 AS5350XM-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 AS5350XM 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 AS5350XM 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 AS5350XM 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 AS5350XM 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.
MGCP
MGCP 1.0 is a protocol for centralized control of VoIP calls by external call-control elements known as media gateway controllers (MGCs) or call agents. MGCP is described in RFC 3435: Media Gateway Control Protocol (MGCP) Version 1.0, published by the IETF.
Package Types
An MGCP call connection involves a series of events and signals-such as off-hook status, a ringing signal, or a signal to play an announcement-that are specific to the type of endpoint involved in the call. MGCP groups these events and signals into packages. A trunk package, for example, is a group of events and signals relevant to a trunking gateway; an announcement package is a group of events and signals relevant to an announcement server.
The Cisco AS5350XM Universal Gateway supports the following MGCP package types:
• Trunk package
• Generic media package
• Dual tone multifrequency (DTMF) package
• DTMF trunk package (for channel-associated-signaling [CAS] endpoints)
• Multifrequency operator services package (for CAS endpoints)
• Multifrequency Wink Start and Immediate Start package (for CAS endpoints)
• Real-Time Transport Protocol (RTP)
• FXR package for fax transmissions
• Announcement server package
• Script package
• Resource Reservation Protocol (RSVP) package (QoS)
Standards-based T.38 Fax Relay and RFC 2833 DTMF Relay are available with MGCP as well as improved voice-quality metrics.
Voice Quality
The extensive voice and fax capabilities of the Cisco AS5350XM Universal Gateway can help build a reliable, high-quality VoIP network. Voice-quality tests confirm that the Cisco AS5350XM Universal Gateway delivers end-to-end voice-quality performance that meets the high standards established for toll-quality voice services in the PSTN. Comprehensive voice-quality testing is a critical component in the Cisco AS5350XM Universal Gateway development process. Cisco conducts subjective voice-quality tests to determine mean opinion scores using a methodology derived from ITU-T Recommendations P.830 and P.831. Objective voice-quality tests are also conducted using the Perceptual Evaluation of Speech Quality (PESQ) algorithm (P.862), an enhanced perceptual measurement for voice quality in telecommunications specifically developed for end-to-end voice-quality testing under real network conditions.
The high-performance design of the Cisco AS5350XM Universal Gateway minimizes delay and packet loss during voice encoding and packetization processes. The Cisco AS5350XM High-Density Packet Voice/Fax Feature Card (AS5X-FC) and DSP Module (AS5X-PVDM2-64) optimize packetization performance and reduce delay up to 20 percent compared with earlier-generation DSP feature cards. Cisco QoS features, including IP Precedence, RSVP, Weighted Fair Queuing (WFQ), Weighted Random Early Detection (WRED), and Multichassis Multilink PPP (MMP) fragmentation and interleaving, implemented on both the universal gateway and backbone routing infrastructure, can provide a low-latency, high-reliability path for sensitive voice traffic through today’s networks.
Echo control is essential for packet-switched networks to carry voice traffic successfully. The Cisco AS5350XM Universal Gateway conforms to the voice tests of ITU-T Recommendation G.168 2002 for echo cancellation with a tail length up to 64 ms. Fixed and adaptive jitter buffering and comfort-noise generation further enhance voice quality. The Cisco AS5350XM also supports the enhanced measurements and call-specific debug features in Cisco IOS Software. In addition, the user can set items such as IP-side attenuation down to the individual T1 or E1 voice port.
Voice Codecs
The Cisco AS5350XM Universal Gateway offers multiple codecs to meet interoperability, compression, and latency requirements for a variety of voice applications. The Cisco AS5350XM High-Density Packet Voice/Fax Feature Card (AS5X-FC) provides complete flexibility in channel allocation to achieve highest densities. Each voice/fax feature card supports from one to six high-density packet voice DSP modules (AS5X-PVDM2-64), providing scalability from 64 to 384 channels. Enabling voice activity detection (VAD) reduces packet traffic through the network. With VAD enabled, the Cisco AS5350XM detects silence and stops transmitting packets when callers stop speaking. Variable frame sizing provides further control over speech packetization.