Search results

Через ваш сервер звонки не проходят. Скорее всего он сконфигурирован на использование внутренних SIP аккаунтов, потому что использовать наши тестовые не получилось. Пришлите пожалуйста на ту же почту аккаунты, которые вы используете для тестирования или SSH доступ к этому тестовому серверу.

Ссылка на вашего тестового клиента, чтобы мы могли проверить будет у нас эхо или нет при созвонах.

В документации раздел не пустой. Просто нужно кликнуть по иконке книги слева чтобы дерево раскрылось или скачать PDF. Правильно ли я понял, что вы используете клиента со стандартными настройками? Сбросте ссылку на logs@flashphoner.com. Мы проверим. При тестировании на других ОС эхо есть...

Какого клиента используете для звонков? Мы тестировали двухсторонние созвоны с нашим стандартным клиентом http://flashphoner.com/webrtc-sip-web-phone-demo/ Эхо не обнаружили. Т.е. берем для теста два ноутбука Win8 и Win7, на одном из них запускаем Chrome, IE или FF, на втором запускаем IE...

Sorry it was updated at the WebRTC Recording thread. Just try latest available builds at http://flashphoner.com/wcs4#download

Howard, We have been testing latest build from here: http://flashphoner.com/wcs4#download From our tests audio and video in VLC is synchronized. Please check.

By the way, we added basic WebRTC stream recording feature: record_webrtc_streams=true in flashphoner.properties config to get this working. WCS will record video to WCS_HOME/records directory in WebM container. Currently recording conains video only. We are going to add audio soon.

It works for me with the latest builds. Make sure rtsp_server_enabled=true in server.properties config I'm able to play stream via VLC: rtsp://192.168.1.5:554/may6OY2dWeXc5HDeQs41J0r1Ln2mIx where 192.168.1.5 - IP address of WCS4 server

Only way to speed this up is to purchase a corresponding Enterprise license. We guarantee work of all features within the Enterprise license and we fix issues ASAP. But I'm afraid it does not fit your requirements 'it is not to much $$$'. Anyway ,we will have a window after a couple of weeks...

Burak, We have investigated the logs. We could not find a root cause by this logs. Is there way to check this SIP call with your SIP credentials? If so, please send two testing SIP accounts to the same address logs@flashphoner.com. We will try to reproduce it with your SIP server. Brief...

Atthachments are not currently supported out of the box. Sure you can try to attach a file and encode it as a text, for example using Base64 then unpack this text on receiver side. The best way to keep message history is to keep them on your Web server. To do that you can use REST API hooks...

Before the test, make sure tcpdump is installed and works properly. yum install tcpdump

Hi Please create a bugreport: 1. Set <push_log>true</push_log> in the client flashphoner.xml 2. Set client_dump_level=2 in the server flashphoner.properties 3. Restart server service webcallserver restart 4. Reproduce your issue and disconnect client (close browser window). Then please zip...

The feature WebRTC as RTSP is not so popular. So we did not have a chance to investigate it due our internal schedule. I believe we will able to do it bit later. Maybe when the issue will collect some votes or when we will have a window in our schedule. If we have a progress here, I will update...

Just to keep issues list 1. Audio / Video synch issue for WebRTC as RTSP feature. 2. User1 - audio call, User2 answers with video, no video on User1 (Chrome - to - Chrome WebRTC)

We are able to reproduce it with latest build. 1. User1 - Video Call - User2 User2 answers with video -- passed (two way audio, two way video) 2. User1 - Video Call - User2 User2 answers with audio only -- passed (two way audio, video playback on User2) 3. User1 - Audio Call - User2 User2...

The latest server build 1200 allows OnCallEvent.sipMessageRaw parameter pass. Therefore you will able to get whole SIP INVITE request at JavaScript layer. See OnCallEvent in our Developer Guide and Call Flow docs. Pass of the parameter is disabled by default due security reasons. Most...

Please clarify your test case. We have tested similar case with our default web-phone example: examples/phone/Phone.html 1) 10001 presses 'Video Call' and calls to 10002. 2) 10002 presses 'Answer'. 10001 - no video playback because 10002 answered with audio only. 10002 - has video playback...

Нужна очень хорошая пропускная способность от сервера конечным пользователям. Поток 640x480 занимает примерно 800kbps. Т.е. 1000 потоков потребуют около гигабитного канала. Желательно 2 CPU, 4-6 ядер каждый. Но скорее всего хватит и одного CPU если нет транскодинга. Транскодинга не будет если...

We are able to reproduce it now. In our tests voice goes little ahead of video in the VLC player. I'll update this thread once we have any news about this issue.