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SIP
lengxingfei
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What is the difference between tag and branch-id?
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转载 2006-02-16 16:14:00 · 826 阅读 · 0 评论 -
Available Telephone Service Features
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转载 2006-02-16 16:36:00 · 702 阅读 · 0 评论 -
What is PTT? What is PoC? How are these related to SIP?
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原创 2006-02-06 10:41:00 · 831 阅读 · 0 评论 -
Basic wireless/3GPP
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转载 2006-02-06 10:49:00 · 801 阅读 · 0 评论 -
What is the relationship between the From, Contact, Via and Record-Route/Route headers?
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转载 2006-02-16 16:12:00 · 837 阅读 · 0 评论 -
IPv6协议在各操作系统下的安装与配置
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转载 2006-02-14 15:32:00 · 1086 阅读 · 0 评论 -
SIP Call Transfer and Call Forwarding Supplementary Services(cisco)
<!-- Id:samkarkh@cisco.com : Date : 10/09/2003Comment : Blockquotes are very important for indentation -->Table Of ContentsSIP Call T转载 2006-02-20 10:47:00 · 3926 阅读 · 0 评论 -
对DTMF在VOIP中应用的研究汇总
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转载 2006-04-10 20:11:00 · 1992 阅读 · 0 评论 -
用OPTIONS方法实现Keep Alive
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转载 2006-04-10 20:17:00 · 1375 阅读 · 0 评论 -
IPv6 Study Resources (Cisco Doc CD)
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)转载 2006-04-13 21:07:00 · 774 阅读 · 0 评论 -
浅说IMS
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,实现转载 2006-04-10 19:53:00 · 890 阅读 · 0 评论 -
SIP Info DTMF
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转载 2006-04-12 11:59:00 · 2059 阅读 · 0 评论 -
MWI = Message Waiting Indicator
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转载 2006-04-21 10:32:00 · 2195 阅读 · 0 评论 -
Message-Waiting Indication (MWI)
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原创 2006-04-21 11:09:00 · 2624 阅读 · 0 评论 -
Codecs payload and voice quality
Codecs payload and voice qualityFrom SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia Table of contents showTocToggle("show","hide")转载 2006-03-20 16:57:00 · 991 阅读 · 0 评论 -
Do 3GPP/3GPP2 use SIP?
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原创 2006-02-06 10:39:00 · 794 阅读 · 0 评论 -
Any relation between SIP URL and IP address?
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原创 2006-02-06 10:24:00 · 729 阅读 · 0 评论 -
VoIP 相关 RFC
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原创 2005-08-24 10:52:00 · 1126 阅读 · 0 评论 -
一篇关于RTP介绍比较全的文章
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原创 2005-08-24 11:00:00 · 5991 阅读 · 1 评论 -
SIP]SIP系列标准导航公告板
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原创 2005-08-24 10:54:00 · 878 阅读 · 0 评论 -
OnDo SIP Server NAT traversal feature
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原创 2005-08-29 18:14:00 · 2363 阅读 · 0 评论 -
【转贴】了解和掌握下面几个命令将会有助于您更快地检测到网络故障所在,从而节省时间,提高效率。
转载 2006-01-04 13:10:00 · 710 阅读 · 0 评论 -
what is SIP Outbound proxy ?
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原创 2006-01-16 18:55:00 · 2625 阅读 · 0 评论 -
Can a proxy fork a non-INVITE request? If yes, what happens if it gets multiple responses?
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原创 2006-01-16 19:02:00 · 683 阅读 · 0 评论 -
What is a dial plan ?
A: A dial plan is a set of rules used for determining whether a complete set of numbers has been entered for the IP原创 2006-01-17 19:15:00 · 1032 阅读 · 0 评论 -
基于p2p的sip电话系统-
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架构支持基本的原创 2006-01-18 17:45:00 · 4855 阅读 · 0 评论 -
what is iLBC and Features ?
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原创 2006-01-17 17:01:00 · 723 阅读 · 0 评论 -
"call progress tones" or "ring back"?
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原创 2006-01-17 19:08:00 · 819 阅读 · 0 评论 -
When the RTP port is set to 8005, why is 8006 used by RTP and 8007 for RTCP for all communications ?
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原创 2006-01-17 19:16:00 · 760 阅读 · 0 评论 -
RTP Silence Suppression
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原创 2006-01-17 17:07:00 · 1145 阅读 · 0 评论 -
开通网络通信-软件技术圈子,欢迎加入
开通网络通信-软件技术圈子,欢迎加入http://blog.csdn.net/group/networkcomm/原创 2006-10-02 16:23:00 · 1256 阅读 · 0 评论