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Sipml5 vs jssip

I see that it is, but I'm still sort of stuck because I can't seem to get a call to work. Three working samples to start with: Use the SIPml5 web phone package on Debian and add it to a virtual host, browse to /sipml5-web-phone Use wget to fetch a copy of the JsSIP sample page from tryit. ; Set these options in repro. ISBN. JSSIP — легковесная Javascript либа для работы с SIP. The world's first HTML5 SIP client (WebRTC). Changes 6 Attacking SIP/VoIP Servers Using VIPROY VoIP Pen-Test Kit for Fun & Profit - Duration: 53:06. Nov 4, 2013 sipML5: an open source JavaScript SIP client; jsSIP: JavaScript SIP library The sipML5 developers have also built the webrtc2sip gateway. Media JavaScript libraries such as JSSIP, SIPML5, PJSIP, and so on, these libraries cater to the SIP/IMS (IP  Apr 7, 2019 OnSIP is happy to introduce SIP. X Versión 1. io - Xirsys. Hola David, Como leiste en el articulo las versiones de Asterisk por encima de la 11. js file to hard-code your SIP proxy address QoffeeSIP is another alternative - this email gives details how ⬤ Codec War (H. 0\VC. 0-beta1 option needs doc clarification vs transport option connection from JsSIP or SIPML5 generate a segmentation - Popular examples are jsSIP, sip-js, QoffeeSIP, or sipML5 • Call Control API - proprietary signaling scheme based on more traditional web tools and techniques - “standard” APIs enhanced to include WebRTC support • Other alternatives based on XMPP, JSON or foobar - Popular examples are jsSIP, sip-js, QoffeeSIP, or sipML5 • Call Control API - proprietary signaling scheme based on more traditional web tools and techniques - “standard” APIs enhanced to include WebRTC support • Other alternatives based on XMPP, JSON or foobar FreeNode #freeswitch irc chat logs for 2014-02-27. There are open source JavaScript libraries (SIP. En septembre de la même année, un canvas logiciel à base de JavaScript pour faire tourner le protocole SIP baptisé JsSIP est lancé par Versatica, équipe déjà à l'origine du brouillon de travail sur les WebSockets [78]. com hosted server infrastructure! Instant free video conferencing The interest in integrating real-time and real-time multimedia communication features into IP networks and its applications, the web including, started a development effort, which brings several SIPML5 client by dubango. Otras Implementaciones SIPML5 World Wide SIP 36. https://segmentfault. sipml5可以用以下链接进行测试: Если у вас есть желание детальнее разобраться в процессах настройки и обеспечения комплексной безопасности локальной и сетевой инфраструктуры, построенной на базе ОС Linux, рекомендую познакомиться с онлайн-курсом 百问 FreeSwitch (第二版) 余洪涌 编著 2014 年 9 月 中国厦门 百问 FreeSwitch(第二版) 第 2 页 文档历史: 版本号 日期 描述 1. 7xxx (principalmente) H. En cada episodio hablaremos acerca del mundo del desarrollo del software. 두 가지를 모두 지원하는 유료 라이브러리는 Plivo, Twilio websdk입니다. LINK_OPTS_0 = $(linkdebug) msvcrt. These issues probably deserve a blog post of their own, but they are not insurmountable. It takes advantage of SIP and WebRTC to provide a fully featured SIP endpoint in any website. WebRTC enables Real-Time Communications (RTC) audio/video capabilities in Web browsers and other devices such as smartphones. If you want to write tables from two – five at the same time and using pencils and papers we need at least four writing hands ( four people), four pencils and four papers one for each to write a table. 711 y Opus Codecs de vídeo Serie G. 1 2013-06-11 yhy 补充windows下的PJSIP软电话和安卓下软电话ImsDroid 的编译和单机最大支持多少线并发通话 1. TOOLS32 = E:\Program Files\Microsoft Visual Studio 10. jssip. LINK_OPTS_0 = $(linkdebug) msvcirt. JsSIP API (V)World Wide SIP 35. JsSIP API (V)World Wide SIP 34. e. Yate SIP phone. IMPROVEMENTS OF JSSIP LIBRARY AS. sipml5 which implements client SIP stack and use WSS as transport, If I get node_webrtc & jssip in node to work SIP is systematic investment plan : This is basically setting up investments in a mutual fund scheme at a fixed frequency ( daily, monthly or quarterly) on a fixed date with a fixed amount for a fixed period. SIP. Vx/ O R LTD S OLICI TUD RESPUES TA Web Browser Web Browser 29. 323 (principalmente) Sin definir Medios RTP/RTCP RTP/RTCP Codecs de voz G. js, JsSIP, sipML5). If you need something more robust then you might have a look at mizu webphone. 264 Seguridad de los medios SRTP/TLS/IPsec SRTP Fuente: ramonmillan. . May. puhuri: I guess there is no other option except recompiling fot get rid of "[INFO] mod_enum. Doubango. SIP sobre WebSocket en el lado del Servidor ¿Por qué es necesario? Tutorial Overview. Enjoy the real integration of SIP within the Web and communicate with SIP networks out there. Get started quickly []. pptx STUN Server Bob Alice STUN Server STUN Server disco disco offer and candidates answer and candidates … connectivity checks … Vanilla ICE as per RFC 5245 WebRTC STUN Server Bob O/A with host or no cands … more cands & conn checks … disco Slide 6 Alice disco 7. QH17472265; ssni; 定期存单破了,又忘记了密码想马上取钱怎么办? ps4 4. Email Patches; Plain Diff; Replace JsSIP by the SIPml5 parent 6e525bcb. Time. Webrtc Client using Jssip - No audio both ways using Free switch and chrome visual-studio visual-studio Release Summary asterisk-13. Commit Score: This score is calculated by counting number of weeks with non-zero commits in the last 1 year period. This in-depth comparison of jssip. Signaling vs Media. 264 vs. In this article we show you how to build a signaling service, and how to deal with the quirks of real-world connectivity by using STUN and TURN WebRTC is a free, open project that enables web browsers with Real-Time Communications (RTC) capabilities via simple JavaScript APIs. JsSIP. And there's a bunch of easy things they can be dropped into your web page to make this happen. VP8 (MTI TBD - IPR discussion) • Media codecs are negotiated with SDP (for now at least) • Requires Secure RTP (SRTP) – DTLS • Requires Peer-2-peer NAT Ninguna Categoria; ESCUELA SUPERIOR POLITÉCNICA DEL LITORAL Tesis de Grado. Now Proceed to configure sipjs/sipml5. js · sipml5 – World's first HTML5 SIP client from Doubango; JsSIP – Written by the  Собственно, выбор скорее между sipml5 и jssip. low level approach  Tonight I have tried two WebRTC clients (JsSIP and sipML5) with Asterisk 11 and get both of them working — echo test calls Perfect results (SIP TCP vs UDP). 1 vs. Audio/video Sessions between different set of peers (Chrome vs. T. Otras Implementaciones SIPML5 World Wide SIP 35. 0. But if you wants use one IP and 443 port then you will try > configre NGINX to proxy all reuests line "/fs-socket/" to From msc at freeswitch. sip related issues & queries in StackoverflowXchanger. net joseluis. Kapanga SIP softphone . uminho. Browse files Options. VP9) ⬤ Many different emerging approaches for Signaling ⬛ JSON over WebSocket (Fall-back to BOSH, COMET) ⬛ SIP over WebSocket ⬛ etc… ⬤ WebSocket is not yet implemented in every HTTP proxy. JsSIPAPI (IV)World Wide SIP 33. Asterisk will be configured to support a remote WebRTC client, the sipml5 client, for the purposes of making calls to/from Asterisk within a web browser. There's a sipML5, which is a way to talk to various standard SIP devices, Phono, and what we're going to show you now, a widget from Zingaya to make a phone call. From these libs you will have a high-level API to work with it, so there is no much VoIP knowledge needed. Commit 498c9655 authored Jan 14, 2015 by XivoBuilder. SIP sobre WebSocket en el lado del Servidor ¿Por qué es necesario? Use’Cases’ • WebRTC’enables’innovave ’use’cases’on’theWeb – WebRTC’It’s’not’meant’tobe’ thenewWeb Telephony’ SIP WebRTC가 클라이언트 JsSIP, sipJS 같은 오픈 소스 libs와, sipml5; SIP-플래시 클라이언트 RED5, 플래시 phoner. JsSIP API (II)World Wide SIP 32. js Xampps(網路視訊與電話 第1頁) Un blog del programa de podcast "Programar es una mierda". THE MOST RELIABLE . sipML5. js makes it easy to utilize WebRTC's APIs and set up SIP communication sessions. Windows Operating system SIP software Xlite is well known SIP softphone for windows dessktop. 168. ⬤ Desktop sharing, statistics is not yet implemented in every browser JsSIP APIWorld Wide SIP 31. 1. Programing with sipML5 API: The API is designed with love to make it easy to develop rich and robust HTML5 applications in few lines of code. Проблему с тишиной удалось решить тем, что мы отказались от STUN в SIPml5. Why did we ultimately decide to fork off from JsSIP? We wanted a  Apr 23, 2014 SIPML5, http://sipml5. you must set the local-network-acl rfc1918. Web RTC. 亲测可以使用,需要freeswitch开启ws 5066端口才可以用,需要用火狐浏览器,其他的浏览器测试不能使用,不能使用https链接,学习足够了,商业也可以使用,可以继承在crm上,非常不错,web电话条,jssip案例,jssip软 亲测可以使用,需要freeswitch开启ws 5066端口才可以用,需要用火狐浏览器,其他的浏览器测试不能使用,不能使用https链接,学习足够了,商业也可以使用,可以继承在crm上,非常不错,web电话条,jssip案例,jssip软 Доброго времени суток всем! Я уже писал о своем опыте работы с WebRTC тут, но учитывая то, что в последнее время всё больше статей на эту тему появляется на хабре и то, что я давно хотел This document defines a set of ECMAScript APIs in WebIDL to allow media to be sent to and received from another browser or device implementing the appropriate set of real-time protocols. Clients * SIP. It surely won’t be long until a full-fledge SIP Client is available in the browser Integrating Asterisk with WebRTC - ground up. js is capable of voice and audio communications, text-based messaging, and data transfers, among other features. Communication. 7系统下载; 综合考虑; oppor9锟截碉拷 Temasys is a Singaporean startup who live and breathe WebRTC. Asterisk and sipml5 interoperability Note: this page is unmaintened and could contain incorrect information. org might explain which of these two domains is more popular and has better web stats. 我在局域网环境中工作. js * sipml5 – World's first HTML5 SIP client * JsSIP – Written by the authors of RFC 7118 and OverSIP. #freeswitch IRC Archive. webrtc2sip is a smart and powerful gateway using RTCWeb and SIP to turn your browser into a phone with audio, video and SMS capabilities. That may be your best choice if you are working in small scale and quite  4 июн 2014 работы с SIP. lib. Всё это чем-то неуловимо напоминает apache vs nginx. Free online heuristic URL scanning and malware detection. JSSIP – MIT license SIP phones in Ubuntu ( Linux system) SFL phone. 263, H. org (Michael Collins) Date: Thu, 31 Jul 2014 14:41:33 -0700 Subject: [Freeswitch-users] How to get conference call status? 最新文章. millan@frafos. com/a/1190000018621781 2019-03-22T22:01:16+08:00 2019-03-22T22:01:16+08:00 Carl https://segmentfault. With. A veces las cosas no salen bien, mejor tomárselo con humor. com - rtc. To cope with network address translators (NATs) and firewalls. It's free to sign up and bid on jobs. They’re acutely aware of the potential of this amazing technology, but also know that only Opera, Firefox and Chrome users can take advantage of it. . Diverses applications sur l'internet utilisent les outils proposés par WebRTC. And one thing that people want to talk from WebRTC is phones. Для браузеров, которые не поддерживают WebRTC, планируется использовать  Library JsSIP sipML5 QoffeSIP SIP-js Microsoft presents an alternative solution, (SRTP- conferencing and collaboration solution" WTC 2014; World SDES vs. 编译器索要的LINK运行库不同,原本以为可以改为msvcrt100. 广西壮族自治区图书馆春节上班又加班费吗; 亲,有谁家孩子在京师阳光幼儿园上学吗?怎么样啊? 山推sl50w装载机 2017-09 there is some movement in this space KeePassXC deserves testing KeePassXC is a community fork of KeePassX, a native cross-platform port of KeePass Password Safe, with the goal to extend and improve it with new features and bugfixes to provide a feature-rich, fully cross-platform and modern open-source password manager. JsSIPAPI (IV)World Wide SIP 34. Fatih Ozavci 26,737 views JsSIP APIWorld Wide SIP 30. Search for jobs related to Sdp android api or hire on the world's largest freelancing marketplace with 15m+ jobs. One thing that had me thinking in particular was about the high level vs. 我的问题如下:我没有从WebRTC获得音频到WebRTC客户端. There are also Library JsSIP sipML5 QoffeSIP SIP-js. Install the repro SIP proxy using the packages from Debian or another Linux distribution like Fedora or Ubuntu. 1、概述nnn2、SIPML5参数设置nnn3、SIPML5、WebRTC信令交互 JsSIP demo JsSIP - 提供的一个兼容WebRTC的JS SIP库,原来托管在github上的一个demo,现在原项目地址似乎不可用了,备份一个。 FreeSwitch 在Windows系统上的安装与运行 SipML5 - The world's first open source HTML5 SIP client. The WebRTC components have been optimized to best serve this purpose. 711 por defecto para la comunicación con websockets. Building WebRTC Apps with JsSIP José Luis Millán jssip. X. Linphone. 2014. com VP8 25. JsSIP allows any website to get real-time communication features using audio and video. org Contact World's first HTML5 SIP client This is the world's first open source ( BSD license ) HTML5 SIP client entirely written in javascript for integration in social networks (FaceBook, Twitter, Google+), online games, e-commerce websites, email signatures 亲测可以使用,需要freeswitch开启ws 5066端口才可以用,需要用火狐浏览器,其他的浏览器测试不能使用,不能使用https链接,学习足够了,商业也可以使用,可以继承在crm上,非常不错,web电话条,jssip案例,jssip软电话,jssip源码,sip软电话源码,sip网页软电话 在与Astrisk争夺WebRTC几周之后,我决定将我的问题放在这个论坛上. Tonight I have tried two WebRTC clients (JsSIP and sipML5) with Asterisk 11 and get both of them working — echo test calls with ulaw (g711u) codec works, but with one-way audio if I call from WebRTC to the SIP softphone. com/u/wangdd 16 <p>最近面试了三个 sipML5:开源JavaScript SIP客户端; jsSIP:JavaScript SIP 我用的vs版本是vs2017professional版本,并未安装所有的工具 在编译kbengine源码 SipML5使用Chrome的实验功能WebRTC实现媒体功能,并用JavaScript封装了一个完整强大的javascript SIP/SDP stack 完成信令的管理,传输层通过Websocket与服务端Gateway通信接入SIP Server,最终通过Video TAG播放视音频内容。 立即下载 • A browser-embedded media engine • Best-of-breed echo canceler • Video jitter buffer, image enhancer • Audio codecs – G. JsSIPAPI (III)World Wide SIP 32. A. ----- 主機 -Elastix 2. So if 26 weeks out of the last 52 had non-zero commits and the rest had zero commits, the score would be 50%. 大家在看. Un blog del programa de podcast "Programar es una mierda". Reliability. WebRTC vs Voip Característica Voip WebRTC Señalización SIP y H. Alexa - Sipjs Competitive Analysis, Marketing Mix and Traffic Log in Does anyone know of a SIP client that can open a web page to a configurable URL when someone calls? We have a web-based database, and I'd like to search by phone number whenever we get an incoming Does anyone know of a SIP client that can open a web page to a configurable URL when someone calls? We have a web-based database, and I'd like to search by phone number whenever we get an incoming You can start with open source projects such as SIPML5 or JsSIP. ソフトフォンを使わずにWeb内で通話する方法 Showing 1-3 of 3 messages JsSIP is a library for the programming language JavaScript. If talking to clients both inside and outside the N. Xlite new version. Real. 情况 - 使用独立的Asterisk服务器从JSSIP调用JSSIP(同一客户端). In theory, you can deploy a SIP server using an open source softswitch (FreeSWITCH, Asterisk) project and purchase "SIP trunking" service to obtain phone numbers and route calls to/from the PSTN. 265 vs. net and sipml5. If behind N. What today works (i. 1'. This feature is not available right now. 将TOOLS32修改为你的VS2010路径. If you want you can use Opus codec for high audio quality. 1449371876" See other formats 链接地址 百问FreeSwitch:VOIP,软交换,FreeSwitch实用案例解答 余洪涌著 作者介绍: 余洪涌:国家系统分析员,从1997年起一直从事语音在电信行业应用方面计算机系统研发工作。 Release Summary asterisk-13. WebRTC - A Future Without SIP? Published on February 12, 2015 February 12, Additionally, open source frameworks like JSSip or sipML5 are enabling the encoding of SIP messages to Javascript Technology enthusiast that has a career in networking and network infrastructure. Please try again later. org/docgen/symbols/SIPml. Using this API, it will be a piece of cake to write HTML5 VoIP applications. 问题从6001(JSSIP)到6002(JSSIP)进行调用时根本没有音频. QuoffeSIP. 5 no funciona el codec Opus, por eso recomiendo no instalarlo y utilizar el G. X Jssip. Milena lmoberdori wuo o I maul: Q. com For clients to exchange metadata to coordinate communication: this is called signaling. libraries, JsSIP [19] and sipML5 [20]. As I am no longer involved in the telco business I won’t update this article anymore. Been involved with the Maemo OS since 2008, and then went on to work with MeeGo and now cover everything to do with the Tizen OS. Sin el soporte de ICE practicamente no podriamos usar ninguna API de WebRTC para Asterisk: JSSIP o SIPML5 de modo que es necesario recompilar Asterisk para que estas APIs puedan funcionar correctamente. Tiempo sin servicio Como consecuencia para poder ofrecer una alta disponibilidad del servicio se deben tomar consideraciones para la planificacin, diseo e implementaciones de las redes en s misma, entre las principales caractersticas con las que se podemos categorizar una red para VoIP se encuentran [30]: Retraso WebRTC. using libraries like JsSIP , and SIPML5, I started to have a change of heart. Windows  . JsSIP implements the SIP WebSocket transport. com Open Tok platform - Vidyo. Comparing JsSIP vs SipML5 may also be of use if you are interested in such closely related search terms as jssip or sipml5, sipml5 vs jssip and jssip vs sipml5. net is tracked by us since June, 2013. Get traffic statistics, SEO keyword opportunities, audience insights, and competitive analytics for Sipjs. 一段时间后,一些RTP数据包正在发送, ICE vs. JsSIP API (II)World Wide SIP 31. config: Video, Chat, and Data Demo. sipml5 已經有一段時間, 也看過 jssip 這套 JavaScript SIP Library. - Muballii Mite woo 0 I‘ 1 Tour Focal 2015 en C. Telestax WebRTC client. SIPJS with flash network support. c:1191 Timezone reloaded 530 definitions" and still see all other INFO messages? FreeNode #freeswitch irc chat logs for 2014-03-21. js. I believe I have mis-read some earlier info from FreeSwitch, which is why I thought a=crypto is not allowed. 2 2013-06-15 yhy 补充使用 sipp 进行对 FreeSwitch 进行压力 WebRTC vs Voip Característica Voip WebRTC Señalización SIP y H. In no time at all, you can have two separate users talking to one another. Trickle ICE Slide from: trickle-ice-iet86-orlando. No need to know how SIP work to start writing your code. 711, Opus are MTI • Video codecs – H. js, our fork of the JsSIP JavaScript library. Then, you can configure a WebRTC SIP client to use your server. 2 INDICE Capitulo I - Instalación de Asterisk Preparación del VPS SSH y clave RSA Clave RSA en Windows Clave RSA en Linux Configuración servidor SSH Utilidades, librerías, dependencias DAHDI LibPRI Res_fax, GoogleTalk, LibiCAL y SRTP Instalación de Asterisk 30 Capitulo II - Configuración inicial de Asterisk Instalar y configurar un cortafuego Carpetas y 目前在做基于WebRTC的语音和视频终端,语音和视频通话的质量都不错。感谢WebRTC,站在巨人的肩膀上,我们可以看得更远。 DECLARAÇÃO Nome: André Manuel Rodrigues da Silva Endereço Electrónico: pg17619 <at> alunos. Contribute to DoubangoTelecom/sipml5 development by creating an account on GitHub. net and edit the provided custom. Anuncio 1 Libro Asterisk 11. Problem plotting satellite data from TDSCatalog python python-xarray metpy python-siphon Updated May 11, 2019 22:26 PM jssip 、sipml5 都是这种解决方案。 通过转换网关实现协议的转换,从而互通。一个开源的网关项目就是 webrtc2sip。 webrtc2sip是一个功能很完善的网关,既实现了信令层,也实现了媒体层,编码转换功能很强大,也可以直接当做媒体网关,用于编解码,沟通两端的 sip related issues & queries in StackoverflowXchanger. sipML5 with Chrome Stable) may not work tomorrow once Chrome implements WebRTC specs in a more strict way (SRTP-SDES support will be removed from Chrome in favour of SRTP-DTLS), and thus traying to make JsSIP to work in the media plane with Asterisk today is a painful race. Tips. c:876 ENUM Reloaded" and "switch_time. xml to the public IP address of your FreeSWITCH. make sure to set the ext-sip-ip and ext-rtp-ip in vars. 4Node. 2014-03-21 (SipML5 AND JsSIP) i get the following message "NO candidate ACL defined Threads usage in C programming. This tutorial demonstrates basic WebRTC support and functionality within Asterisk. Over the time it has been ranked as high as 445 199 in the world, while most of its traffic comes from USA, where it reached as high as 343 363 position. html . VP8) future (H. Scan websites for malware, exploits and other infections with quttera detection engine to check if the site is safe to browse. WebRTC: JSEP Approach. Several JavaScript SIP stacks are being developed, such as sipML5 (‘The world’s first open source HTML5 SIP client’) and the older, also open source SIP-JS project. WebRTC  Como biblioteca usaremos ou sipML5 ou jsSIP. So, they’ve released Skylink, a free plugin for OS X and Windows which brings WebRTC to Safari and Internet Explorer. pt Nº do Bilhete de Identidade: Título da Dissertação: Novas Arquiteturas Web para aplicações Disponibilidad del Servicio vs. Basicamente o ramal será criado usando sua API Rest, o registro será realizado pela  With specific needs and functionality in mind, our development team decided to fork the open project JsSIP to suit the contours of GetOnSIP and more to come. auto, and prefix the ext-sip-ip and ext-rtp-ip to autonat:X. 0 2012-11-10 yhy 建立文档 1. I'm calling a local extension on my FreeSwitch server, 7779, which currently just plays a voice prompt. To check out the full code for all three demos, click the button below. This is pure SIP on the web (no protocol conversion, no limits). org Fri Aug 1 01:41:33 2014 From: msc at freeswitch. 0 option needs doc clarification vs transport option of connection from JsSIP or SIPML5 generate a segmentation 2016-12-24 12:05 GMT+02:00 Sergey Safarov : > Hello guys > I want configure user frendly WebRTC server based on FreeSwitch and SipML5 > client. js allows you to utilize WebRTC’s APIs using just JavaScript. 要照著範例來安裝測試, 連線都有問題. SIP UA registration with the help of SIP Web  2018年4月4日 之前几篇文件介绍了freeSWITCH 和WebRTC 结合在一起需要的各种环境,现在 到了最关键的一篇,使用 JsSIP 来创建一个DEMO 。这次我们需要  Jul 1, 2019 Nginx and what I found worked was to proxy to the actual IP address (192. Smart SIP and Media Gateway to connect WebRTC endpoints. JsSIPAPI (III)World Wide SIP 33. Clients. EORLD Signaling vs Media SessionDescription < Caller Callee SessionDescription 28. > > It can be easy done in FreeSwitch and NGINX is bounded to different > IP/ports. !WebRTC services and platforms - Tokbox. lib,但没找着 Full text of "OReilly. Mode of the spam folder can be set via new API variable G_SpamFolder [*] 2012-09-19: [SV-1178] Config - c_accounts_policies_login_revealpasswords does not show plain password anymore [-] 2012-09-19: [SV-1374] SMS - statistics and config handling reworked to better comply with remote console mode [-] 2012-09-19: [SV-41] System - Account cache vs All tests succeed with Chrome now [*] 2014-03-11: SIP Media Proxy - Final To: tag used after 200 OK, JsSIP uses different ;tag for 100 Trying and SIP dialog [+] 2014-03-11: SIPMediaProxy - RTPDump support [-] 2014-03-11: [SV-5253] SMTP - Smart Attach: Problem with dot/double dot processing fixed [*] 2014-03-10: RTPPacketLogger class - free of WebRTC What’s going on and is it of use to NRENs Mihály Mészáros, NIIF Institute eduCONF Workshop 13/03/14 在freeswitch开放ws后,要使用webrtc去对接,主流还是simpl5和jssip. sipml5 vs jssip

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