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These tones usually accompany the 183 SESSION PROGRESS status produced and sent by intermediate VoIP gateways. The client receives such tone signals and plays the as usual audio stream. Unfortunately, for now you can't turn this off.

Web Call Server works as a stream video server. Therefore you can develop a conference based on WCS.

1. You can implement support for conferences in Flashphoner without mixing. In this case the conference is straight exchange of WebRTC streams many-to-many between each other. 2. operation in networks with bad connection. This is all about quality real-time delivery of the video from a 3G device...

1. Yes. 2. No. 3. Yes, it is possible with Flashphoner Web Call Server. You can send streams and play streams from other web-clients. Based on this functionality, you can implement just any logic you need.

Regarding the second option: Let's say we have a SIP video conference. There are 2 users from web-pages (powered by Flashphoner) and 2 SIP-devices via hardware or software SIP-phones. There's also a mising server. Each participant of the conference makes a video call to this mixing server. Think...

There are two types of conferences: 1. when each SIP device plays N-1 streams and sends N-1 streams, where N is totla number of SIP devices connected to the conference. This can be implemented, but it will take long and the cost will be pricey too. 2. When each SIP device plays and sends just...

Hello. Most likely, sound is missing because IP addresses are incorrectly configured in flashphoner.properties config. Here are the correct settings: ip=<external ip>, ip_local=<local IP address of the server> In addition, you should route UDP ports UDP [30000-32000]. It should work with these...

Simply deactivate your license on your initial server, then activate it again on the second server using the activation script bin/activation.sh

Hello. It seems you should open ports on the server. Take a look: http://flashphoner.com/what-firewall-and-nat-rules-should-be-used-with-wcs/ This example shows how to configure ports for Amazon EC2. Besdies you should set flashphoner.properties ip=<external ip address of the server>...

WebRTC Media & Broadcasting Server умеет получать исходный поток через RTP и ретранслировать на браузеры с поддержкой WebRTC: Chrome, Firefox, Opera - без установки стороннего ПО, и на иные браузеры с установкой плагина WebRTC. Исходный поток должен быть доступен через сообщения RTSP signaling...

Здесь вы можете найти черновой вариант документации, включая пошаговую инструкцию о том, как настроить простой видеочат без SIP-сервера: http://wcs-docs.flashphoner.com/display/WCS32/WebRTC+Video+Chat+With+Flash+Fallback Руководство еще будет дорабатываться. Попробуйте установить клиент на своем...

Flashphoner WebCallServer supports not just Flash calls, but also WebRTC stack making possible high-quality calls in Chrome and Firefox without using Flash. Our solution was tested with major IP-PBX (Asterisk, FreePBX) and SIP-Proxy. All functionality for the call button (Click2Call) is already...

Sure, this is possible. The license costs $260 per one line (one browser calling). You can ask sales and licensing questions to sales@flashphoner.com."

As I understand it, you mean a call service from a website. for that purpose you need to store a database of companies - end users of the service - and supply them with a code to embed to a website. In this case you should apply the autologin mechanism and Click2Call. Please refer to these links...

По такой схеме можно работать, учитывая следующие моменты: 1. "Конференция" может звонить на обычный телефон через серверное API Flashphoner 2. Каждый пользователь может воспроизвести аудиопоток обычного телефона через Wowza, т.к. "Конференция" знает имя потока. Flashphoner разработан на трех...

Есть код клиентского wowza приложения, совместимый с Wowza4: https://github.com/flashphoner/flashphoner_client/tree/rsgw-wz4 Это опенсорс, его можно интегрировать на разных уровнях: flash, javascript, server-side

Browsers: Desktop: latest versions of Chrome, Firefox, Opera, IE, Safari Mobile(Android): latest versions of Chrome, Firefox iOS doesn't support WebRTC and Flash in Safari and Chrome

Try settings add_register_auth_headers=true in flashphoner.properties, the registration should go ok then.

You need to edit flashphoner.properties: rtp_activity_detecting=true,60 In this case, if there's no voice from the SIP within 60 seconds, the call hangs up. The default value is 5 seconds.

Пожалуйста, вышлите ваш ключ на почту helpdesk@flashphoner.com. Посмотрим, что можно сделать.