sip

SIP协议 会话发起协议(一)

六月ゝ 毕业季﹏ 提交于 2019-11-27 03:20:11
会话发起协议(SIP)是VoIP技术中最常用的协议之一。它是一种应用层协议,与其他应用层协议协同工作,通过Internet控制多媒体通信会话。 SIP - 概述 以下是有关SIP的几点注意事项 - SIP是用于通过因特网协议创建,修改和终止多媒体会话的信令协议。会话只不过是两个端点之间的简单调用。端点可以是智能电话,笔记本电脑或可以通过因特网接收和发送多媒体内容的任何设备。 SIP是由IETF(Internet Engineering Task Force)标准定义的应用层协议。它在 RFC 3261中 定义。 SIP体现了客户端 - 服务器体系结构,以及使用 HTTP 和URL的URL和URI 以及 SMTP 的文本编码方案和头样式。 SIP采用SDP(会话描述协议)的帮助,它描述了用于通过IP网络传送语音和视频的会话和RTP(实时传输协议)。 SIP可用于双方(单播)或多方(多播)会话。 其他SIP应用包括文件传输,即时通讯,视频会议,网络游戏,以及流多媒体分发。 下图说明了SIP在一般方案中的适用性 - 通常,SIP协议用于两个或多个端点之间的互联网电话和多媒体分发。例如,一个人可以使用SIP发起对另一个人的电话呼叫,或者有人可以与许多参与者建立电话会议。 SIP协议的设计非常简单,配置有限的命令。它也是基于文本的,所以任何人都可以读取SIP会话中的端点之间传递的SIP消息。

sip 注册流程

我是研究僧i 提交于 2019-11-27 03:19:57
基本注册流程示意图: 注册流程描述如下: 1、 SIP代理向SIP服务器发送REGISTER请求; 2、 SIP服务器向SIP代理发送响应401,并在响应的消息头WWW-Authenticate字段中给出适合SIP代理的认证体制和参数; 3、 SIP代理重新向SIP服务器发送REGISTER请求,在请求的Authorization字段给出信任书,包含认证信息; 4、 SIP服务器对请求进行验证,如果检查出SIP代理身份合法,向SIP代理发送成功响应200 OK,如果身份不合法则发送拒绝服务应答。 注册信令消息示范: Register sip:SIP服务器编码@目的域名或者IP地址端口 SIP/2.0 Via:SIP/2.0/UDP 源域名或者IP地址端口 From:<sip:SIP设备编码@源域名>;tag=185328220 To: <sip:SIP设备编码@源域名> Call-ID:ms1214-322164-681262131542511620107-0@172.18.16.3 CSeq:1 Register Contact: <sip:SIP设备编码@源IP地址端口> Max-Forwords:70 Expires: 7200 Content-Length: 0 SIP/2.0 401 Unauthorized To: sip:SIP设备编码@源域名 Content

Minimum SDP for making a H264 RTP stream?

随声附和 提交于 2019-11-27 01:57:49
问题 I'm looking for an example of a minimum necessary SDP for setting up a H264 video stream.| The assumption is that the receiver can play H264 as long as it gets the required parameters through SDP. I have found a related document herehowever it uses lots of optional parameters in the examples, and I'm looking for the bare required minimum. 回答1: Here is the bares minimum SDP. It is a file called test.sdp which has the following content: c=IN IP4 10.5.110.117 m=video 5004 RTP/AVP 96 a=rtpmap:96

SIP-Client for Raspberry Pi that works from command line?

≡放荡痞女 提交于 2019-11-27 00:58:20
问题 i want to use my raspberry pi as a SIP/VOIP-Phone, just controlling the RPI via SSH. I found some tutorials and it seems that Twinkle is one of the most useful apps for that. So i successfully installed twinkle on my RPI, one SPI-Client on my Android-Phone and for know i am able to send text messages from one to another. The thing is: I have to use the Twinkel GUI with X-Server-forwarding (currently using MacOS with X11 and iTerm). But i kind of want to automate the whole process, like using

SIP library for iOS with non-GPL license

给你一囗甜甜゛ 提交于 2019-11-27 00:40:42
问题 Are there any non-GPL SIP libraries/SDKs that'll let me implement SIP for iOS? 回答1: These are the ones I know about: Proprietary: vaxvoip Open source: GPL/LGPL sofia-sip linphone exosip 回答2: oSIP is LGPL and therefore can be used (as a library) in commercial and closed-source projects. It compiles easily for arm. It is the SIP library used by linphone for iPhone; however, linphone is GPL. 回答3: These days SIP over WebRTC is getting popularity. It uses SIP Signaling and WebRTC as Media which is

How to integrate Linphone into an existing project (SIP in IOS)

早过忘川 提交于 2019-11-27 00:38:05
问题 I am looking for integrate Linphone library in my ios project. And I dont find tutorial to help me. This project offers very little documentation, but seems to work well. For information, here is the project page which provides its open source code : Linphone open source code I already compiled sources by following all steps and this work very well, But the next step is to integrate a part of code in my personal iPhone project for use SIP layer, with the management of audio and video. In the

Failed to create SipSession; network unavailable

我只是一个虾纸丫 提交于 2019-11-26 23:37:42
问题 I've loaded the Android SIP Demo into Android Studio. I registered an account on getonsip.com, added it to my phone accounts, and verified it works by calling the phone from the getonsip.com webapp and also from one Android phone to another. All good so far. I modified the Java code from the demo slightly to accommodate the getonsip settings: SipProfile.Builder builder = new SipProfile.Builder(username, domain); builder.setOutboundProxy(proxy); builder.setPassword(password); builder

How to use pjsip on android device in order to have G.729 codec functionality

时光怂恿深爱的人放手 提交于 2019-11-26 21:04:40
问题 I want to use G.729 audio codec on my android application. I have did a lot research on this and came to know that pjsip is most promising solution for this. But I have not much idea about all this. can someone provide me complete steps for using pjsip in existing android application and how can i include support of G729 codec via pjsip . Any help will be appreciated. EDIT : Here is my android.mk file. I want to know that,have i did this right? and how to use those C functions in my java code

C# SIP Stack/Library [closed]

假如想象 提交于 2019-11-26 17:06:32
I am looking for a good SIP library either written in C# or that provides a C# wrapper. Does not necessarily need to be free. Has anyone used anything good? To clarify, I am talking about the VoIP protocol SIP. I am really looking for more than a google search on this one. I would like to someone to come back and say "We use ______, and it works great." To clarify further, I am looking for a library that would help me to implement a SIP proxy server without having to manually construct the SIP packets. I'm not looking for asterisk, freeswitch or any other pbx software. I am looking to create a

Android SIP stack - what to use? [closed]

拥有回忆 提交于 2019-11-26 15:38:27
问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 3 years ago . I need to create use an SIP stack on Android, which will work with asterix and will give users the possibility to change codecs (i need to implement G729 and some other codecs). I'm new in this field (voice and codec), and every bit of information will be appreciated. In my research across the Internet, I found