lengxingfei的专栏

Branch IDs allow proxies to match responses to forked requests. Without them, a proxy wouldnt be able to tell which branch a response corresponds to. Tags, in To headers, are of no help here since th

Automatic RecallWhen you push *69, an announcement will come on telling you the number of the last incoming call and the time it came in. By simply pushing "1" you can instruct the system to call this

PTT stands for Push To Talk. This is the telephony technology which simulates walkie talkie type of communication. It has become very popular in the wireless arena. Typically when people hear this t

Basic wireless/3GPPQuestionsWhat’s the difference between GPRS and UMTS?What’s the relation between GPRS and IP?Can you do voice calls with GPRS?What is the relation between GPRS and SIP?W

All these headers determine how requests and responses are routed in a network of SIP proxy servers. Roughly, the distinction is: - From: Used for subsequent requests from the callee to the caller if

IPv6协议在各操作系统下的安装与配置- -Tag： IPv6协议                                          2005-06-28 14:39:46　 作者： 　来源：赛迪网本文将为大家介绍如何在Redhat Linux 9操作系统、Windows 2000 Server操作系统、Windows XP Professional操作系统和Free BS

  <!-- Id:samkarkh@cisco.com : Date : 10/09/2003Comment : Blockquotes are very important for indentation -->Table Of ContentsSIP Call T

简介双音多频DTMF（Dual Tone Multi-Frequency）信令，逐渐在全世界范围内使用在按键式电话机上，因其提供更高的拨号速率，迅速取代了传统转盘式电话机使用的拨号脉冲信令。近年来DTMF也应用在交互式控制中，诸如语言菜单、语言邮件、电话银行和ATM终端等。 由于DTMF在传统通信领域中的广泛使用，所以在VOIP中，DTMF仍是发挥着重要的作用。 一个DTM

转自noiile@hotmail.com在SIP应用中，假如用户在用SIP UA享受某种计时收费的业务，比如在线观看收费电视，突然死机，这时服务器不能得知任何关于用户设备死机的信息，计费服务仍在继续。这显然是种不合理的情况，服务器必须实时的知道用户的状态，这种情况presence服务器也不一定能帮助解决问题。因为用户的SIP UA只有在改变状态的情况下才发PUBLISH消息到Presenc

IPv6 (Internet Protocol Version 6)IPv6 Switching ServicesIPv6 RoutingIPv6 Services and ManagementIPv6 Broadband AccessNAT Protocol Translation (NAT-PT)IPv6 Tunnel ServicesIPv6 QoS (Quality of Service)

转自 noiile@hotmail.com 概述IMS（IP Multimedia Subsystem）是3GPP在Release 5版本提出的支持IP多媒体业务的子系统，它的核心特点是采用SIP协议和与接入的无关性。在网络融合的发展趋势下，3GPP、ETSI和ITU-T都在研究基于IMS的网络融合方案，目的是使IMS成为基于SIP会话的通用平台，同时支持固定和移动的多种接入方式，实现

The SIP INFO method can be used by SIP network elements to transmit DTMF tones out-of-band as telephone-events in a reliable manner independent of the media stream. In the DTMF relay method the body o

MWI = Message Waiting Indicator Usually an audio or visual signal that a voicemail or other type of message is waiting. Examples: Stutter dial-tone Flashing light on phone Vibratio

Message-Waiting Indication (MWI)             2006-4-21      MWI is a common feature of telephone networks and uses an audible indication (such as a special dial tone) that a message is waiting

Codecs payload and voice qualityFrom SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia Table of contents showTocToggle("show","hide")

Yes they do. 3GPP adopted SIP in 2001. Your humble slaves were involved in the discussions that preceded that critical decision (i.e. preferring it on H323). 3GPP came up with the IMS (IP Multimedia

Not really. SIP URL is an application layer Identifier, similar to an email or web address. It is associated with a person not a computer. IP address is associated with a computer (see the IP Basics

VoIP 相关 RFC RFC3261 SIP: Session Initiation Protocol 这可说是VoIP SIP相关标准中最重要的一个，取代了旧版的SIP标准RFC2543 下载： http://www.faqs.org/ftp/rfc/pdf/rfc3261.txt.pdf中文版本 http://www.boyinsoft.com/cn-rfc3261.pdfRFC 3262

RTP, Real-time Transport ProtocolDescriptionGlossaryRFCsPublicationsObsolete RFCsDescription:Protocol suite:TCP/IP.Type:Application layer protocol.Port:5004 (UDP).SNMP MIBs:iso.org.dod.internet.mgmt.m

Core SIP DocumentsRFC 2543SIP: Session Initiation Protocol (obsolete) RFC 3261SIP: Session Initiation Protocol SDP Related DocumentsRFC 2327Session Description Protocol (SDP) RFC 3264An Offer/Answer M

OnDO SIP Server NAT traversal feature OnDO SIP ServerOSS A -  An OnDO SIP Server with on global IP address.OSS B -  An OnDO SIP Server running on a machine that has two NIC cards installed.One of the

SIP Outbound proxy - the way out From the SIP RFC: Outbound Proxy: A proxy that receives requests from a client, even though it may not be the server resolved by the Request-URI. Typically, a UA i

Yes, a proxy can fork a non-INVITE request. However, it must forward only a single response upstream, 200 or otherwise. Thus, only a single 200 is ever forwarded upstream. This is in contrast to INVIT

A: A dial plan is a set of rules used for determining whether a complete set of numbers has been entered for the IP

摘要p2p系统天生拥有高扩展性、健壮性和高容错性的特点，这些特点得益于系统没有中央服务器并且网络是自己管理的这种结构。本系统实现了在 p2p系统中较长的延迟的代价下定位感兴趣的资源。internet 话可以被看作一个p2p架构的应用，它在一部分和另外一部分定位和通讯时在p2p这种自组网上实现。我们的目的是构建一个基于SIP信令的纯p2p架构的ip电话系统。 我们的p2psip架构支持基本的

iLBC is a VOIP codec originally created by Global IP Sound but made available (including its source code) under a restricted but free and fairly liberal license, including permission to modify. Quoted

The SIP server being called, such as an Internet telephony gateway,can return any number of provisional status messages that indicate callprogress. Typically, this is just 100 (Trying) followed by 180

RFC requires the RTP port to be even numbers and the next RTP port to be RTCP port numbers (an odd number). So the ATA always forces the RTP port to even numbers. If you select 8005 it will use both 8

Endpoints sending audio as an RTP stream are not required to send packets during silent periods. The capability to stop sending RTP packets during silent periods is known as "Silence Supression" or VA

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