SIP Gateways[sip的网关服务器介绍]

SIP网关是一种应用,用于将SIP网络与使用其他信号协议的网络对接。它终止信号路径,并可能终止媒体路径,如SIP到H.323网关。SIP到PSTN网关则同时终止信号和媒体路径,实现SIP与PSTN之间的通话。网关可以分解为媒体网关(MG)和媒体网关控制器(MGC),它们分别管理媒体连接和呼叫控制协议。此外,网关通常支持大量用户,而不像用户代理那样仅支持单个用户。

    SIP Gateways
A SIP gateway is an application that interfaces a SIP network to a network utilizing another signaling protocol. In terms of the SIP protocol, a gateway is just a special type of user agent, where the user agent acts on behalf of another protocol rather than a human. A gateway terminates the signaling path and can also terminate the media path, although this is not always the case. For example, a SIP to H.323 gateway terminates the SIP signaling path and converts the signaling to H.323, but the SIP user agent and H.323 terminal can exchange RTP media information directly with each other without going through the gateway. An example of this is described in [H.323 to SIP Call]

A SIP to Public Switched Telephone Network (PSTN) gateway terminates both the signaling and media paths. SIP can be translated into, or interwork with, common PSTN protocols such as Integrated Services Digital Network (ISDN), ISDN User Part (ISUP), and other Circuit Associated Signaling (CAS) protocols, which are briefly described in [PSTN Protocols]. A PSTN gateway also converts the RTP media stream in the IP network into a standard telephony trunk or line. The conversion of signaling and media paths allows calling to and from the PSTN using SIP. Examples of these gateways are described in [SIP to PSTN Call Through Gateway]  and Figure 3.1: shows a SIP network connected via gateways with the PSTN and a H.323 network. There is work underway to standardize the SIP/H.323 interworking function .

Click To expand
Figure 3.1: SIP network with gateways.

In the figure, the SIP network, PSTN network, and H.323 networks are shown as clouds, which obscure the underlying details. Shown connecting to the SIP cloud are SIP IP telephones, SIP-enabled PCs, and corporate SIP gateways with attached telephones. The clouds are connected by gateways. Shown attached to the H.323 network are H.323 terminals and H.323-enabled PCs. The PSTN cloud connects to ordinary analog black telephones (so-called because of the original color of their shell), digital ISDN telephones, and corporate private branch exchanges (PBXs). PBXs connect to the PSTN using shared trunks and provide line interfaces for either analog or digital telephones.

Gateways are sometimes decomposed into a media gateway (MG) and a media gateway controller (MGC). An MGC is sometimes called a call agent because it manages call control protocols (signaling), while the MG manages the media connection. This decomposition is transparent to SIP, and the protocols used to decompose a gateway are not described in this book.

Another difference between a user agent and a gateway is the number of users supported. While a user agent typically supports a single user (although perhaps with multiple lines), a gateway can support hundreds or thousands of users. A PSTN gateway could support a large corporate customer, or an entire geographic area. As a result, a gateway does not REGISTER every user it supports in the same way that a user agent might. Instead, a non-SIP protocol can be used to inform proxies about gateways and assist in routing. One protocol that has been proposed for this is the Telephony Routing over IP (TRIP) protocol [6], which allows an interdomain routing table of gateways to be developed. Another protocol called Telephony Gateway Registration Protocol (TGREP) [7] has also been developed to allow a gateway to register with a proxy server within a domain.

在VoIP通信中,SIP(Session Initiation Protocol)和H.323是两种常见的协议,它们分别用于建立、管理和终止多媒体通信会话。由于这两种协议的结构和功能存在差异,因此在需要将SIP协议转换为H.323协议的场景中,通常会使用SIP到H.323网关解决方案。 ### SIP到H.323网关的功能 SIP到H.323网关的主要功能包括: - **协议转换**:将SIP协议的消息格式和会话控制机制转换为H.323协议的消息格式和会话控制机制。 - **媒体流转换**:处理音频和视频流的编码/解码,确保两种协议之间的媒体流可以互通。 - **NAT穿越**:支持NAT(网络地址转换)穿越功能,以确保在复杂的网络环境中能够正常通信。 - **安全性支持**:提供H.235安全协议支持,包括密码认证和媒体加密。 ### 常见的SIP到H.323网关解决方案 1. **开源解决方案**: - **GNU Gatekeeper (GnuGk)**:GnuGk是一个H.323网守,支持跨平台运行,并且可以与其他协议(如SIP)集成。它支持多种数据库系统,并提供NAT穿越功能[^2]。 - **Yate**:Yate是一个多协议软电话客户端,支持H.323、SIP和IAX协议。它可以作为SIP到H.323网关使用,适用于Linux、Solaris、Windows和各种Unix系统[^1]。 2. **商业解决方案**: - **Cisco Unified Border Element (CUBE)**:Cisco的CUBE设备支持SIP到H.323的协议转换,并提供高质量的媒体处理和安全性功能。 - **Avaya Session Border Controller (SBC)**:Avaya的SBC设备支持SIP和H.323之间的互操作性,并提供NAT穿越和媒体加密功能。 3. **硬件网关**: - **Dialogic Gateways**:Dialogic提供了一系列硬件网关,支持SIP到H.323的转换,适用于企业级和运营商级应用。 - **AudioCodes Media Gateways**:AudioCodes提供高性能的媒体网关,支持SIP和H.323之间的转换,并提供丰富的媒体处理功能。 ### 配置示例 以下是一个简单的SIP到H.323网关配置示例,假设使用Yate作为网关: ```python # Yate配置文件示例 [sip] enabled = yes port = 5060 [h323] enabled = yes port = 1720 [protocol_conversion] sip_to_h323 = yes h323_to_sip = yes ``` 在上述配置中,Yate启用了SIP和H.323协议,并启用了协议转换功能,使得SIP和H.323之间的通信成为可能。 ### 选择网关的考虑因素 - **协议兼容性**:确保网关支持SIP和H.323之间的完整协议转换。 - **性能**:根据预期的通信负载选择合适的网关性能。 - **安全性**:确保网关支持必要的安全功能,如媒体加密和身份验证。 - **可扩展性**:选择支持未来扩展的网关,以便在需要时增加更多的功能或容量。 ###
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