sip

SIP学习之旅【环境搭建篇】

本秂侑毒 提交于 2020-01-17 01:14:51
SIP学习之旅【环境搭建篇】 经过两天的资料收集对SIP有了初步的了解,在网络电话VOIP应用方案中很多都是采用Linux服务器部署电信级SIP Server软件或者使用专门的硬件服务器来提供SIP Server服务,客户端用SIP电话机或软电话,结构组成包括:SIP终端,代理服务器,重定向服务器,注册服务器器,鉴权服务器,SIP Server服务器等,很庞大、很复杂、很无奈(没有多余的电脑给我装Linux,更没有SIP硬件),所以我需要的是Windows下运行的SipServer和SoftPhone,以及一个抓包工具就可以了。 SipServer :MiniSipServer ,下载地址 http://www.myvoipapp.com/download/index.html SoftPhone : X-Lite ,下载地址 http://www.counterpath.com/x-lite-4-for-windows-download.html 抓包工具 :MiniSniffer(这工具有个缺陷就是没有报文时间,不过用习惯了就是它了), 点击下载 SIP协议文档: SIP流程图带动画解释.pdf, 点击下载 组网结构图: 部署: 1. 安装并启动MiniSipServer,一路Next就行,通过开始菜单启动,默认监听 UDP 5060 端口;系统自带三个默认用户

SIP简介

孤街醉人 提交于 2020-01-17 01:08:56
一、SIP简介 1.1 SIP简介   SIP(Session Initiation Protocol) 是一个应用层控制协议,它能建立、 修改和终止多媒体会话。SIP是由IETF(Internet Engineering Task Force)的 MMUSIC(Multi- party Multimedia Session Control)工作组提出。   目前SIP协议标准是RFC3261。SIP协议还是正在快速发展的协议.SIP(Session Initiation Protocol) 是一个应用层控制协议,它能建立、修改和终止多媒体会话。  SIP是由IETF(Internet Engineering Task Force)的MMUSIC(Multi- party Multimedia Session Control)工作组提出。目前SIP协议标准是RFC3261。  SIP协议还是正在快速发展的协议. 1.2 SIP特点及优点 SIP采用文本编码,使用请求/响应模式 建立会话和描述会话分离 端到端的协议 互操作性 可扩展性 1.3 SIP用途 SIP作为一个创建服务的平台 --- SIP最重要的用途 SIP可以整合Internet服务与多媒体服务,并能传送到用户真实位置。 SIP可以统一多种通信格式。 SIP很适合传输即时消息及状态信息。 SIP使用现有的框架提供新的服务

libeXosip2(3-1) -- eXosip2 INVITE and Call Management

☆樱花仙子☆ 提交于 2020-01-17 01:04:02
eXosip2 INVITE and Call Management SIP messages and call control API Functions int eXosip_call_set_reference (struct eXosip_t *excontext, int id, void *reference) void * eXosip_call_get_reference (struct eXosip_t *excontext, int cid) int eXosip_call_build_initial_invite (struct eXosip_t *excontext, osip_message_t **invite, const char *to, const char *from, const char *route, const char *subject) int eXosip_call_send_initial_invite (struct eXosip_t *excontext, osip_message_t *invite) int eXosip_call_build_request (struct eXosip_t *excontext, int did, const char *method, osip_message_t **request

Are the “:” or “/” in SDP values for attributes tokens by themselves with spaces optionally allowed around them?

雨燕双飞 提交于 2020-01-16 10:29:26
问题 In SDP, colon(':') and slash('/') are used in many attribute values (both the standard and a= extensions). Here are just a few of them: b=AS:41 a=rtpmap:96 AMR-WB/16000/1 a=fmtp:96 mode-change-capability=2; max-red=80 I want to know (both for parsing and generating SDP), if space is allowed around them. All examples point to not having a space around them. I think Section 9 of RFC 4566 which gives the grammar for SDP is not clear about this. 回答1: I would say, that usually, SDP don't like

How to send SIP message using Python sockets

扶醉桌前 提交于 2020-01-13 20:49:33
问题 I need to send SIP message using Python sockets, I've made that client sends something to server but I'm not able to make that client sends a SIP message INVITE to server #!/usr/bin/python import socket R_IP = '192.168.2.1' R_PORT = 5060 message = 'INVITE sip:user1110000000350@.com SIP/2.0 To: <sip:user4110000000350@whatever.com>\x0d\x0aFrom: sip:user9990000000000@rider.com;tag=R400_BAD_REQUEST;taag=4488.1908442942.0\x0d\x0aP-Served-User: sip:user4110000000350@whatever.com\x0d\x0aCall-ID:

How to send SIP message using Python sockets

◇◆丶佛笑我妖孽 提交于 2020-01-13 20:48:29
问题 I need to send SIP message using Python sockets, I've made that client sends something to server but I'm not able to make that client sends a SIP message INVITE to server #!/usr/bin/python import socket R_IP = '192.168.2.1' R_PORT = 5060 message = 'INVITE sip:user1110000000350@.com SIP/2.0 To: <sip:user4110000000350@whatever.com>\x0d\x0aFrom: sip:user9990000000000@rider.com;tag=R400_BAD_REQUEST;taag=4488.1908442942.0\x0d\x0aP-Served-User: sip:user4110000000350@whatever.com\x0d\x0aCall-ID:

Android 2.3 using external JAIN-SIP(J-SIP) Stack | Classpath

你。 提交于 2020-01-13 18:28:43
问题 I'm trying to use the JAIN-SIP (J-SIP) library on Android 2.3.3. (http://jsip.java.net/) Therefore I utilized a very basic SIP-example app that comes with the SIP-Stack. This example runs perfectly on Android devices <= Android 2.2. The problem ist, that Android 2.3.3 brings a SIP-Strack with it's system-libraries, which is a slightly modified version of the JAIN-SIP stack. So on Android 2.3.3 devices, the internal SIP-Stack is loaded (because of same package/library names) instead of the

About Notification for Missed Call in android

萝らか妹 提交于 2020-01-13 07:20:16
问题 I am using voip sip for calling processing calls when Push-Notification received. I want to create top bar notification for missed call . I am already processing calls when Push-Notification receives. I already searched lot of things but nothing helped me. My question is, i'm making calls after receiving Push-Notification , so likewise i want to display notification after receiving Push Notification, so for that how do I differentiate for which i am receiving Push-Notification . 回答1: The

Video Call through SIP in Android

本秂侑毒 提交于 2020-01-13 03:18:05
问题 I am beginner in Android and working on an application which could make video call over IP using SIP. I searched about it a lot on Google and also on StackOverflow and all I got is that I would need some Native Library, about which I don't have so much knowledge. I looked after Linphone and Csipsimple and IMSdroid but didn't get anything (means I don't need the working application, I just want to know how they work). Using inbuild SIP stack in Android, I am able to make Audio Calls but don't

Understanding how SIP, WebRTC and PSTN work together [closed]

∥☆過路亽.° 提交于 2020-01-12 05:35:27
问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 5 years ago . I have been learning more about WebRTC, SIP and PSTN and how they work together especially the ability to receive phone calls in browser. I have spent some time on Twilio's website and I like the way they are able to make you receive a phone call From PSTN to your browser. I want