Tryit Jssip Git

Le 15 avril 2015, un BarCamp de 2H sur le VoIP open source a été organisé sur IRC, entre des communautés en France et Québec. I will be moving these changes into the public GIT repository soon. JsSIP '전화걸기' 기능에서 'User Denied Media Access' ? Asterisk 16. 0, JsSIP no longer includes the rtcninja module. My current setup contains a cisco asa 5505 with two cisco sf300-24p switches, the data and voice vlans have been setup on the switches and I've started to setup the needed vlans on the ASA, but it. This is known as a flat dependency graph and it helps reduce page load. net joseluis. The ice stuff does not like > waiting and I am also not sure if jssip has implemented early media. Any help concerning this matter is greatly appreciated. Starting with version 0. Unfortunately my work stops at this stage as I am not capable to advance any further on this subject/feature. For questions or usage problems please use the jssip public Google Group. Starting from 3. Could you please guide me where can I get the latest code for JsSIP webpage. WEBRTC是一个开源项目,其宗旨是让WEB浏览器通过简单的JavaScript具备实时通信(Real-Time Communications (RTC) )的能力。. This is because some subtle errors may prevent execution of cron commands, eg. cn 码云Gitee 企业版售前及售后使用咨询:400-606-0201. 很多单位在使用内外网通信时,总是要转换编码,但很多编码又是需要版权费用的,我们就是给客户提供正版付了授权费用的转码系统,可软件,可软硬件一体化解决。. 摘要: 今天调试 发现 注册的分机 的 `Auth-User` 居然是 `unknown` !!! 怎么回事? 仔细对比检查 发现, internal profile 指定了 `apply-register-acl` 的参数 ,值为 `domain`, 而默认配置是注释掉这个 参数的, 在看 acl::do阅读全文. Contribute to versatica/tryit-jssip development by creating an account on GitHub. Install Bower. net is much easier to implement although it is not the best however it is well developed and maintained. js is loaded. / home / the Javascript SIP library / Download. 0, JsSIP no longer includes the rtcninja module. 6 연동 테스트를 진행하고 있는데, Session Timers 설정 시, 호가 특정 시간이 지나면 종료되는 증상이 발생하고 있다. JsSIP implements the SIP WebSocket transport. com Mon Feb 1 00:11:12 2016 From: bote_radio at botecomm. x branch, which does include rtcninja. For questions or usage problems please use the jssip public Google Group. js ', ' bower_components/angular-mocks/angular-mocks. Next message: [Freeswitch-users] 488 Not Acceptable Here (INCOMPATIBLE_DESTINATION) with JsSIP and OverSIP + FreeSWITCH Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] Hi, I'm using JsSIP from a webpage to make a SIP call to FS, using OverSIP as a Websocket->SIP proxy. com Mon Apr 1 11:20:10 2019 From: vishalmpai at gmail. One- month trial available. 711 por defecto para la comunicación con websockets. WEBRTC是一个开源项目,其宗旨是让WEB浏览器通过简单的JavaScript具备实时通信(Real-Time Communications (RTC) )的能力。. 1500 (Firefox shows the same behaviour). So now I am gonna integrate JsSIP instead of SIPml5 on the AWS instance and configure to our needs. Starting with version 0. Raspberry Pi and Windows 10 IOT support USB speakers and microphones and the board itself has an audio jack, but it would be nice if Microsoft / Cirrus Logic would support the Cirrus Logic Audio Card daughter board for the Raspberry Pi 2 under Windows 10 IOT. I am not sure if this is just my configuration, but I can get it work in Chrome, but there seems to be about a five second delay between when I click answer and when the call connects in Asterisk. However, the jssip-rtcninja package is based on the 2. 0 100 Trying. Could you please guide me where can I get the latest code for JsSIP webpage. 摘要: 今天调试 发现 注册的分机 的 `Auth-User` 居然是 `unknown` !!! 怎么回事? 仔细对比检查 发现, internal profile 指定了 `apply-register-acl` 的参数 ,值为 `domain`, 而默认配置是注释掉这个 参数的, 在看 acl::do阅读全文. Thanks for contributing an answer to Stack Overflow! Please be sure to answer the question. From chandranraviram at gmail. 0 + JsSIP 3. This is pure SIP on the web (no protocol conversion, no limits). x branch, which does include rtcninja. cn 码云Gitee 企业版售前及售后使用咨询:400-606-0201. I'm trying to set up a webapp using JsSIP 3. This is because some subtle errors may prevent execution of cron commands, eg. Getting Started. Guest User-Public Pastes. Comment by Brian West [ 21/Apr/17 ]. JsSIP exposes the module via the JsSIP. GitHub is home to over 28 million developers working together to host and review code, manage projects, and build software together. reload asterisk JsSIP安装配置. io with https://tryit. WebRTC November 7, 2013 Balatongyörök / Hungary Mészáros Mihály. For questions or usage problems please use the jssip public Google Group. Atlassian Sourcetree is a free Git and Mercurial client for Mac. There are indeed a couple of hardcoded values, which possibly can be either managed as input arguments for the binary, or extracted somehow from the original pcap in a more dynamic fashion. Capitulo VII - Dialplan - Configuración avanzada 160 7. From vishalmpai at gmail. WebRTC What's going on and is it of use to NRENs Mihály Mészáros, NIIF Institute eduCONF Workshop 13/03/14. Bower provides hooks to facilitate using packages in your tools and workflows. cn 码云Gitee 企业版售前及售后使用咨询:400-606-0201. debug accessor. 8b git a549012 2014-02-23 02:18:36Z 64bit on Chrome and FF on Android - this issue appears resolved. js is loaded. Bower is optimized for the front-end. Raspberry Pi and Windows 10 IOT support USB speakers and microphones and the board itself has an audio jack, but it would be nice if Microsoft / Cirrus Logic would support the Cirrus Logic Audio Card daughter board for the Raspberry Pi 2 under Windows 10 IOT. Thanks guys! > > > ----- > Date: Fri, 27 Mar 2015 17:39:55 -0500 > From: anthony. x branch, which does include rtcninja. an asterisk is put after packages in dbs format, which may then contain localized files. JsSIP Library - No request or response for call method Hot Network Questions How can you evade tax by getting employment income just in equity, then using this equity as collateral to take out loan?. Comment by Brian West [ 21/Apr/17 ]. I already uploaded the git code of JsSIP. However, the jssip-rtcninja package is based on the 2. JsSIP Library - No request or response for call method Hot Network Questions How can you evade tax by getting employment income just in equity, then using this equity as collateral to take out loan?. However, the jssip-rtcninja package is based on the 2. cause One value of Failure and End Causes. io settings) by defining a window. org > Subject: Re: [Freeswitch-users] unexpected segfault with latest debian and > libmyodbc. Visit jssip. Multiple JsSIP User Agents can be created (this is useful for having different SIP accounts running in the same web application). Site created with nanoc. All you need is jssip open source that works with webrtc. see #11534. o FreeSwitch的账号管理,路由管理,二次开发时使用esl的李浩录制视频 tags:FreeSwitch Video 创建时间:2016-12-20 11:07:12. Make sure you have added logging to all the possible failure cases catch(err) while accessing webRTC API's. I can help in testing and parameters only from now on!. net joseluis. com Mon Apr 1 11:20:10 2019 From: vishalmpai at gmail. Thanks guys! > > > ----- > Date: Fri, 27 Mar 2015 17:39:55 -0500 > From: anthony. Just point it to wss of asterisk. Now customize the name of a clipboard to store your clips. However, the jssip-rtcninja package is based on the 2. WebRTC What’s going on and is it of use to NRENs Mihály Mészáros, NIIF Institute eduCONF Workshop 13/03/14. d to master, so tryit does use it. Install Bower. 1500 (Firefox shows the same behaviour). Seems to be an issue with lxc, installing/refreshing snaps directly on the host work every time. The other day I was successfully able to get reSIProcate and the basicClient example program to run on my Raspberry Pi 2 under Windows 10 IOT Core! Very few changes were required to the stack in order to get it to compile as an ARM target. Debian Quality Assurance. reload asterisk JsSIP安装配置. For questions or usage problems please use the jssip public Google Group. It worked fine when the server was on public IP but when we put behind firewall and do port forward it was having RTP issue. I'm able to Register successfully, but when I make a call from JsSIP UA to FreeSWITCH, I get a 180 Ringing, but af. Furthermore the STUN server isn’t actually defined anywhere in the jssip tutorials or documentation, You have to dig through their codebase on tryit to actually see what they are doing, which is why FreePBX does this as well (otherwise you’d have issues). io settings) by defining a window. So eventually I tested this with the JsSIP try-it hosted and it worked like a charm with dtls enabled on the freeswitch. W3C CSS3 CSS3. js:21391 JsSIP:Transport received WebSocket text message: SIP/2. JSSIP with Bandwidth Voice API ⚠️ Bandwidth no longer supports WebRTC per rtcpMuxPolicy. Hi, I've been looking at JsSIP 0. UPDATE: with this project, I won a place in the 4th generation of startups of Wayra Mexico. Starting from 3. Clearly it cannot find the package. His library gives you the connection parameters in a set amount of intervals and in a nice presentable way. x branch, which does include rtcninja. If multiple packages depend on a package - jQuery for example - Bower will download jQuery just once. 1449371876" See other formats. I tried the example code on the git repo, with our own asterisk server and it doesn't work, won't make a call. However, the jssip-rtcninja package is based on the 2. New tryit-jssip application. I can find some documentation regarding TURN servers in an old version (0. Building WebRTC Apps with JsSIP José Luis Millán jssip. Getting Started. Capitulo VII - Dialplan - Configuración avanzada 160 7. 7 Limitar llamadas salientes: funciones GROUP y. From bote_radio at botecomm. / home / the Javascript SIP library / Download. I will be moving these changes into the public GIT repository soon. Getting Started. Multiple JsSIP User Agents can be created (this is useful for having different SIP accounts running in the same web application). NGS, or Next Generation Support, is a project that I created to participate in the TADHack event. Full text of "OReilly. Learn how to version file deletions on the command line, stop tracking a file while retaining it on the file system, and how the GitHub d. New tryit-jssip application. Getting Started. Agradecimientos A mi tutor y profesor, D. I already uploaded the git code of JsSIP. JsSIP + OverSIP IITelefonía SIP en TU web: Comunicación entre usuarios web y otros dispositivos SIP Integración PBX y PSTN Telefonía en tu intranet Convergencia de CRM. x branch, which does include rtcninja. 點進去後呢, 頁面長這樣, 左邊是讀者可修改的文字區塊, 右邊則是網頁的結果:. Raspberry Pi and Windows 10 IOT support USB speakers and microphones and the board itself has an audio jack, but it would be nice if Microsoft / Cirrus Logic would support the Cirrus Logic Audio Card daughter board for the Raspberry Pi 2 under Windows 10 IOT. WEBRTC是一个开源项目,其宗旨是让WEB浏览器通过简单的JavaScript具备实时通信(Real-Time Communications (RTC) )的能力。. The app has a settings page, where callstats. js ', ' bower_components/angular-mocks/angular-mocks. 0 connection to a Asterisk server. So now I am gonna integrate JsSIP instead of SIPml5 on the AWS instance and configure to our needs. For bug reports or feature requests open an Github issue. Hi, I've been looking at JsSIP 0. Contribute to versatica/tryit-jssip development by creating an account on GitHub. As per my understanding it is not codec issue as the same codec (opus) is used on dialing out from JsSIP to the same sip end-point and it works. Thanks guys! > > > ----- > Date: Fri, 27 Mar 2015 17:39:55 -0500 > From: anthony. reload asterisk JsSIP安装配置. callstatsjssip(ua, AppID, AppSecret); Demo app. Look under the hood! Explore how Git commands affect the structure of a repository within your web browser with a free explore mode, and some constructed. Starting with version 0. [email protected] Hi, I've been looking at JsSIP 0. For questions or usage problems please use the jssip public Google Group. SETTINGS variable before the tryit-jssip. Following the instructions provided in the github repo, I am getting the following errors on executing 'gulp dev':. js ', ' bower_components/angular-mocks/angular-mocks. net joseluis. but when i make call from jssip to android apprtc , i am not having audio. see #11534. 04 running in lxc containers are having issues installing and refreshing. wow lag 2 sec ago 2 sec ago. Getting Started. 1449371876" See other formats. New tryit-jssip application. I've been working on this for the last few days and I pleased to say that I managed to get through the series of problems that the learning curve entails and now the app is finally working. Seems to be an issue with lxc, installing/refreshing snaps directly on the host work every time. com (Vincent Xia) Date: Mon, 1 Jul 2013 09:32:16 +0800 Subject: [Freeswitch-users] call_timeout and lua in dialplan In-Reply-To: References: Message-ID: got it, thanks a lot!. Q&A for system and network administrators. I checked the order js files where was fine and I'm using JsSip( jssip-0. Мы хотим разработать sip-телефон с библиотекой jssip. 4 sec to load all DOM resources and completely render a web page. net joseluis. I'm trying to set up a webapp using JsSIP 3. WEBRTC简介WEBRTC是一个开源项目,其宗旨是让WEB浏览器通过简单的JavaScript具备实时通信(Real-TimeCommunications(RTC))的能力。WEBRTC目前支持JS和HTML5,项目由Google. com (Vishal Pai) Date: Mon, 1 Apr 2019 16:50:10 +0530 Subject: [Freeswitch-users] Fwd: conference layout in verto In-Reply-To: References: Message-ID: Is it feasible please advice. so > > > We actually recommend Threading=0 for postgres and mysql for the most part. Now customize the name of a clipboard to store your clips. Thanks guys! > > > ----- > Date: Fri, 27 Mar 2015 17:39:55 -0500 > From: anthony. Git repository hosted by Bitbucket. wow lag 2 sec ago 2 sec ago. > > You could compare results with sipml5 and you can also contact the user > groups for both projects on google groups for additional insight. From chandranraviram at gmail. Starting from 3. 點進去後呢, 頁面長這樣, 左邊是讀者可修改的文字區塊, 右邊則是網頁的結果:. For questions or usage problems please use the jssip public Google Group. Getting Started. io with https://tryit. UPDATE: with this project, I won a place in the 4th generation of startups of Wayra Mexico. Scenario is rather simple but I am not sure exactly what is causing JsSIP to send 488. is where it bombs out. However, the jssip-rtcninja package is based on the 2. web软电话 jssip+freeswitch 软电话条 jssip案例 12-02 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. For bug reports or feature requests open an Github issue. One- month trial available. 0, JsSIP no longer includes the rtcninja module. Atlassian Sourcetree is a free Git and Mercurial client for Windows. JSSIP with Bandwidth Voice API ⚠️ Bandwidth no longer supports WebRTC per rtcpMuxPolicy. com Mon Apr 1 11:20:10 2019 From: vishalmpai at gmail. 0), but apparently this feature was removed. Hola David, Como leiste en el articulo las versiones de Asterisk por encima de la 11. jssip-demos / tryit / index. 0, JsSIP no longer includes the rtcninja module. I’ve been working on this for the last few days and I pleased to say that I managed to get through the series of problems that the learning curve entails and now the app is finally working. As described previously, this version uses JSCommunicator (which is based on the popular JsSIP) as the underlying phone framework. stable https://git. 8b git a549012 2014-02-23 02:18:36Z 64bit on Chrome and FF on Android - this issue appears resolved. The other day I was successfully able to get reSIProcate and the basicClient example program to run on my Raspberry Pi 2 under Windows 10 IOT Core! Very few changes were required to the stack in order to get it to compile as an ARM target. I can help in testing and parameters only from now on!. ⬜ SRTP-DTLS (git version) ⬜ video transcoding fs-video branch ⬛ Asterisk ⬜ SIP over Websocket ⬤ SIP Proxy ⬛ Kamailio ⬜ SIP over WebSocket ⬛ OverSIP ⬜ SIP over WebSocket ⬤ RTP PROXY ⬛ mediaproxy-ng ⬤ JS client library ⬛ Doubango SIPML5 ⬛ JsSIP ⬤ Gateway ⬛ Doubango webrtc2sip (GW) ⬤ Web Conferencing ⬛ Big Blue. Es la posibilidad de comunicar utilizando navegadores Web, y no solamente, que implementen las API en JavaScript que permiten a los desarrolladores implementar servicios y aplicaciones que interaccionan con los navegadores mismos. Repository of code using JsSIP. Though we try not to break "normal" usage of our JsSIP fork (i. js and related "tryit" libs (2 days old now) the lastest Freeswitch GIT (2 days old now) Chromium is 28. Older versions of chrome may still work. JsSIP + OverSIPComunicación multimedia entrenavegadores utilizando SIP comoprotocolo de señalizaciónComunicación SIP entrenavegadores y dispositivos SIPconvencionales 34. Thanks for contributing an answer to Stack Overflow! Please be sure to answer the question. Sure, just register into a FS server and make FS call to JsSIP (so JsSIP becomes DTLS active/client). the latest JsSip 0. 1500 (Firefox shows the same behaviour). Also here is another web phone that can be used with asterisk https://tryit. 0), but apparently this feature was removed. No exceptions or errors are shown in the console. Starting from 3. net is the current demo site, with all it's lights and shadows. 5 Las Subrutinas 7. Older versions of chrome may still work. sudo apt-get install build-essential libsqlite3-dev libxml2-dev libncurses5-dev libncursesw5-dev libiksemel-dev libssl-dev libeditline-dev libedit-dev curl libcurl4-gnutls-dev libjansson4 libjansson-dev libuuid1 uuid-dev libxslt1-dev liburiparser-dev liburiparser1 git autoconf libbfd-dev -y Step 3 : Install SRTP stuff. As per my understanding it is not codec issue as the same codec (opus) is used on dialing out from JsSIP to the same sip end-point and it works. Look under the hood! Explore how Git commands affect the structure of a repository within your web browser with a free explore mode, and some constructed. Getting Started. 0, JsSIP no longer includes the rtcninja module. Contribute to Ojero/jssip-demos development by creating an account on GitHub. Unfortunately my work stops at this stage as I am not capable to advance any further on this subject/feature. 4 El contexto Subscribe 7. Thanks for your response. Le 15 avril 2015, un BarCamp de 2H sur le VoIP open source a été organisé sur IRC, entre des communautés en France et Québec. The Session Initiation Protocol, defined in "RFC 3261", is an application layer signalling protocol for establishing and modifying long-running relationships between two or more peers. AngularJS, NodeJS, Gulp, Angular Material, Bootstrap, D3. However, the jssip-rtcninja package is based on the 2. Atlassian Sourcetree is a free Git and Mercurial client for Mac. Full text of "OReilly. d to master, so tryit does use it. I've played a little more and tried the demo at www. 0 connection to a Asterisk server. is where it bombs out. If multiple packages depend on a package - jQuery for example - Bower will download jQuery just once. Thanks for contributing an answer to Stack Overflow! Please be sure to answer the question. Contribute to Ojero/jssip-demos development by creating an account on GitHub. Many people have now heard of the EFF-backed free certificate authority Let's Encrypt. For questions or usage problems please use the jssip public Google Group. Using open public domains (such as SIP2SIP, JSSIP Tryit Server, or sipML5. WebRTC November 7, 2013 Balatongyörök / Hungary Mészáros Mihály. JsSIP + OverSIP IITelefonía SIP en TU web: Comunicación entre usuarios web y otros dispositivos SIP Integración PBX y PSTN Telefonía en tu intranet Convergencia de CRM. New tryit-jssip application. Provide details and share your research! But avoid …. 5 no funciona el codec Opus, por eso recomiendo no instalarlo y utilizar el G. Getting Started. jssip Asterisk con Websockets para WebRTC y probando SIPML5 ATENCIÓN: Este artículo ya no es útil puesto que Chrome en su versión 35 en adelante ha pasado su sistema de encriptación para WebRTC de SDES a SRTP/DTLS como estaba planificado desde principios de Enero 2014. x branch, which does include rtcninja. [email protected] 0 Development (uncompressed code, 564KB): jssip-0. For questions or usage problems please use the jssip public Google Group. wow lag 2 sec ago 2 sec ago. SETTINGS variable before the tryit-jssip. JsSIP User Agent is the core element in JsSIP. JsSIP implements the SIP WebSocket transport. com > To: freeswitch-users at lists. Getting Started. net Website and Documentation jssip. I am using JsSIP library for connecting to my freeswitch SIP server and make a call on it. 1500 (Firefox shows the same behaviour). 1500 (Firefox shows the same behaviour). JsSIP exposes the module via the JsSIP. View git blame. cn 码云Gitee 企业版售前及售后使用咨询:400-606-0201. It represents the SIP client associated to a SIP account. For bug reports or feature requests open an Github issue. see #11534. the latest JsSip 0. I've been working on this for the last few days and I pleased to say that I managed to get through the series of problems that the learning curve entails and now the app is finally working. However, the jssip-rtcninja package is based on the 2. For questions or usage problems please use the jssip public Google Group. io with https://tryit. New tryit-jssip application. 0, JsSIP no longer includes the rtcninja module. Hola David, Como leiste en el articulo las versiones de Asterisk por encima de la 11. So now I am gonna integrate JsSIP instead of SIPml5 on the AWS instance and configure to our needs. 8b git a549012 2014-02-23 02:18:36Z 64bit on Chrome and FF on Android - this issue appears resolved. Now customize the name of a clipboard to store your clips. Site created with nanoc. Starting from 3. 0), but apparently this feature was removed. 1500 (Firefox shows the same behaviour). com (Raviram Chandran) Date: Sun, 1 Oct 2017 13:12:58 +0530 Subject: [Freeswitch-users] How to overwrite Q850 errors code in freeswitch?. x branch, which does include rtcninja. here is my code that is same to sample code on JsSIP github page: jssip_1. However, the jssip-rtcninja package is based on the 2. Fetching contributors…. 2 Pattern Matching 7. Full text of "OReilly. For questions or usage problems please use the jssip public Google Group. W3C CSS3 CSS3. js is loaded. 0 connection to a Asterisk server. More updates to come in the future posts :). io with https://tryit. Git repository hosted by Bitbucket. net joseluis. Anyone with a Power BI license can now log into the account to see how the Power BI REST API works, get information, or perform operation on Power BI artifacts you have access to - without writing any code using the Power BI REST API Try-it Tool. Bower provides hooks to facilitate using packages in your tools and workflows. JSSIP with Bandwidth Voice API ⚠️ Bandwidth no longer supports WebRTC per rtcpMuxPolicy. JsSIP - 提供了一个兼容WebRTC的JS SIP库,在github上有一个用这个库的demo,我们可以到 这里 下载,并直接使用它。. 0 + JsSIP 3. 0, JsSIP no longer includes the rtcninja module. JsSIP '전화걸기' 기능에서 'User Denied Media Access' ? Asterisk 16. io in JsSIP settings. The app allows entering settings via an HTTP form in the Login section. The following simple JavaScript code creates a JsSIP User Agent instance and makes a SIP call: // Create our JsSIP instance and run it: var configuration = {. However, the jssip-rtcninja package is based on the 2. Comment by Brian West [ 21/Apr/17 ]. SETTINGS variable before the tryit-jssip. JsSIP implements the SIP WebSocket transport. com > To: freeswitch-users at lists. Guest User-Public Pastes. net Download As Node git#oschina. Eu estava no meu trabalho belo e suando sangue (como fala meu amigo Francisco. This is because some subtle errors may prevent execution of cron commands, eg. The app allows entering settings via an HTTP form in the Login section. net joseluis. UPDATE: with this project, I won a place in the 4th generation of startups of Wayra Mexico. I'm able to Register successfully, but when I make a call from JsSIP UA to FreeSWITCH, I get a 180 Ringing, but af. the latest JsSip 0. However, the jssip-rtcninja package is based on the 2. Hi, I've been looking at JsSIP 0. Я работаю в телекоммуникационной компании. com Mon Feb 1 00:11:12 2016 From: bote_radio at botecomm. minessale at gmail. 1 Monitoreo de las extensiones remotas con Corosync 7. This is because some subtle errors may prevent execution of cron commands, eg. So now I am gonna integrate JsSIP instead of SIPml5 on the AWS instance and configure to our needs. reload asterisk JsSIP安装 配置 JsSIP - 提供了一个兼容WebRTC的JS SIP库,在github上有一个用这个库的demo,我们可以到 这里 下载,并直接使用它。. Unfortunately my work stops at this stage as I am not capable to advance any further on this subject/feature. o FreeSwitch的账号管理,路由管理,二次开发时使用esl的李浩录制视频 tags:FreeSwitch Video 创建时间:2016-12-20 11:07:12. com Sun Oct 1 07:42:58 2017 From: chandranraviram at gmail. Мы хотим разработать sip-телефон с библиотекой jssip.