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Hello If you provide RTSP url we will setup 2 hours long test to reproduce freeze. Report form

Good day. Plerase clarify, what server hardware configuration do you use and how much memory you allocate for Java heap? Mixer decodes all incoming streams and encodes outgoing stream, so it requires a lot of CPU. We recommend at least 2 CPU cores per one mixer plus at least 2 CPU cores for...

Yes, 183 Session Progress SDP message instructs the caller side to play a media stream according to SDP. So Asterisk should send this message and then RTP media data to the caller. Please check the Asterisk configuration.

Good day. This seems like Chrome bug affecting MP4 metadata parsing, because Firefox, Safari and VLC plays MP4 recording files normally. But we created a ticket WCS-3465 to check if metadata we writing conforms MP4 container specification. If yes, this is definitely a Chrome bug. As workaround...

Good day. We implemented stripCodecs option in iOS SDK (ticket WCS-3416) in build 2.6.78. Please see Media Devices Swift example description for detalis: iOS Media Devices Swift. We still working on multiple streams recording to MKV container in ticket WCS-3419. Will let you know when the...

We checked the dump and tested calls via your Asterisk using SIP accounts provided. Some calls in the dump contains 183 Session Progress SDP packet sent from Asterisk to caller (but not all the calls): In our test with SIP accounts provided, there is no such packet, both in calls from WCS to...

Good day. Please provide a link to RTSP stream using this private form.

Hello Quick solution: chown -R flashphoner:flashphoner /usr/local/FlashphonerWebCallServer/client2 Step-by-step: 1. Backup folder /usr/local/FlashphonerWebCallServer/client2 This folder is old Web SDK content 2. Remove folder client2 and re-create empty folder client2 3. Download and...

Hello Password should be MD5() So if password is admin, then MD5('admin') = 21232f297a57a5a743894a0e4a801fc3 admin: {username: admin, password: 21232f297a57a5a743894a0e4a801fc3, active: true}

Hello Regarding issue WCS-3463. Please check out your /etc/hosts config If your /etc/hosts looks like 127.0.0.1 myfqdn.com Then it won't work because domain must be resolved to public IP address. Therefore we marked issue as "Can't reproduce". Please provide more information how to...

Unfortunately, we received just empty form. If the file size exceeds 30 M please share a link to some cloud storage.

Провели тесты кластера Kubernetes. К сожалению, из-за того, что при описании контейнера нет возможности указать произвольный диапазон портов, WebRTC не работает в контейнерах k8s. Таким образом, CDN в кластере не будет работать: узлы смогут обмениваться только сигналингом через фиксированный...

This topic is about the ClickToCall case. In this case, you usually have only one callee number. So yes, if the callee number has changed, it should be changed in backend script.

Hello Thank you for the report. It looks like a regression. Reproduced on our server. We have created internal issue WCS-3463. Will let you know once it is fixed.

У нас есть официальный образ в докере WCS в Docker, но в Kubernetes мы его не тестировали. Также необходимо учесть, что при развертывании контейнеров в докере для WebRTC (который является многопортовым протоколом, и используемый набор портов достаточно велик, в отличие от RTMP, RTSP или HLS)...

Please also clarify: don't you hear a ringtone when Phone Min example shows RING status? If yes, this is normal behaviour: there is no media traffic yet until call is accepted. So if you need a ringtone, you should customize the example code by adding a ringtone playback start on...

Hello Please make a dump on WCS end: tcpdump upd -s 4096 -w log.pcap This log.pcap file should contain SIP call as listed here https://docs.flashphoner.com/display/WCS52EN/SIP Generally the ring tone is an RTP stream. And we have to check what codec and payload used. Send log.pcap via the...

We tried to make test calls between two browsers via your server. they all fails with the following message in WCS logs: 09:12:53,230 ERROR bstractDTLSTransport - Stun-Clbk-thread udp//95.179.156.48:31414 Close socket after 10 tries 09:12:53,231 ERROR SipCall - Stun-Clbk-thread...

We cannot reproduce the issue with latest WCS build 5.2.1154 on our Asterisk server. Your Asterisk server seems to be unavailable (no response from SIP port both by TCP and UDP). Please check your Asterisk server, it must be available 24/7 for tests. Also, please provide accees to your WCS...

Good day. Unfortunately, we cannot access the server using SSH credentials you've provided. The license key seems to be already activated on some server. So please try to deactivate the license cd /usr/local/FlashphonerWebCallServer/bin/deactivate.sh If this does not help, please contact...