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Good day. We checked the report. If you're using a custom AWS image, please use Amazon Linux 2023 or Ubuntu 22.04 or 24.04 as base OS with OpenJDK 21. If you're using Flashphoner image from AWS marketplace, please update to latest one. We also recommend to use OpenJDK, not Correto JDK. In 5.3...

WCS does not refer to any SIP server configs There was probably an actie SIP session between WCS and Asterisk. By default, session is kept in one hour after client disconnects from WCS. So, if you change some Asterick parameters, tha active sessions may be affected.

Good day. Please make sure UDP ports are available for both sides: Port routing checking If all the ports group are available, and the issue still persists, collect a report including a traffic dump as described and send using this link.

Good day. We fixed a new builds uploading, now both WCS and Web SDK should be available.

Good day. WCS does not support pulling SRT streams from external source. Only SRT publishing is supported. We raised the ticket WCS-4825 to implement this feature and will let you know about progress. Please keep the demo source available for testing.

На прошлой неделе проходил аудит лицензий. Возможно Ваша лицензия была помечена как expired. Пришлите пожалуйста на sales@flashphoner.com 1. Номер лицензии (в админке во вкладке License). 2. Скриншот с последним платежом по этой лицензии.

В прикрепленном логе видно, что истекла лицензия: Убедитесь, что: - сервер лицензий https://my.flashphoner.com/ доступен с вашего сервера; - часы вашего сервера синхронизируются по NTP; - помесячная лицензия оплачена. Возможно также что вы используете триальную лицензию. Срок действия триальной...

Добрый день. Сервер присылает сигналинговое сообщение getVersion с текущей версией сервера, это документировано. Однако на стороне Web SDK эти данные никак не используются, сообщение игнорируется, кроме диагностического вывода в консоль. Завели тикет WCS-4810, добавим минимальную обработку...

You can change the buffer size settings: rtmp_out_buffer_enabled=true rtmp_out_buffer_polling_time=20 rtmp_out_buffer_start_size=10000 rtmp_out_buffer_initial_size=10000 For example, the settings above should give a 10 seconds delay.

Good day. No. The only way to add a delay still is RTMP output buffer. So you should continue with this workaround.

We fixed the issue in build 5.3.254. You should update and set client_timeout=5000 reg_expires=60 sip_force_session_expires=60 In this case, SIP REGISTER messages will be sent every 50 seconds from WCS to Asterisk while browser page is connected. WCS will stop sending registration messages...

Please gather a traffic pcap dump between WCS and PBX TTS. tcpdump -s 4096 -w log.pcap Analyze this dump in Wireshark and make sure G.711 RTP packets are captured by the tcpdump. Once it is captured, please send us using form We will setup internal test with sipp to replay this dump and...

Good day. Please clarify the case: 1. Caller establishes audio only call from browser to the TTS 2. Caller hears audio from the TTS sometimes fast, sometimes slow or 2. Caller hears distorted audio from the TTS (missing phrases, robotic voice etc)? What do you mean: SIP calls between two...

Builds downloading issue is fixed now.

We checked the server and tested a SIP call. The problem is between browser and WCS, not between WCS and 3CX: This occurs at callee side when incoming call is initiating between WCS and Chrome 140 and above. See this thread. The issue was fixed in WCS build 5.3.165. Please update and check...

We'll check and try to fix downloading issue.

Good day. Seems like the problem is between browser and WCS, not between WCS and Asterisk: This occurs at callee side when incoming call is initiating between WCS and Chrome 140 and above. See this thread. The issue was fixed in WCS build 5.3.165. Please update and check. Note that WCS 5.3...

Good day. Please provide SSH access to WCS instance and two SIP accounts to test a call via your PBX using this private form. Or, if this is nt possible, please collect a report including traffic dump and send using this form. We will check and give a recommendations.

Unfortunately, we do not provide ETA for the forum tickets. Please be patient.

Here is how we understand your configuration: 3.34.141.204 Client 119.10.0.1 TURN (3456) + NGINX (443) 119.10.0.2 WCS This is how WebRTC connection should be established: 1. Websocket Signaling 3.34.141.204 > 119.10.0.1 (NGINX reverse proxy) > 119.10.0.2 3.34.141.204 < 119.10.0.1 (NGINX...