Search results

Good day. We raised the ticket WCS-4038 to implement an YUV->RGB->YUV example, will let you know about progress here.

Yes, in this case the ping checking options are not available on top level, so flashphoner.js should be patched. We raised the ticjket WCS-4037 to add the parameters to RoomApi.connect. Unfortunately, yes. Both MacOS and iOS Safari do not support navigator.connection and even...

Good day. You don't need to tweak WebSDK because the options are available when creating a websocket session (GitHub) Flashphoner.createSession({ urlServer: url, receiveProbes: receiveProbes, probesInterval: probesInterval }).on(SESSION_STATUS.ESTABLISHED...

There is no news yet. Usually, we do not provide any ETA on forum issues. Please be patient

If you want just force particular PBX, you can add two lines into config file /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties Then restart server process: systemctl restart webcallserver outbound_proxy = my.pbx.com outbound_port = 5060 Here my.pbx.com - is IPv4 address or FQDN...

Seems like the problem is not in server settings, they are just slightly different. We will investigate it and let you know about progress.

You don't need to set up a new WCS instance to make a calls via another PBX. You can just use a SIP outbound proxy needed as we described above. In this case all WebRTC traffic goes through single WCS instance, but SIP traffic goes through selected PBX. If you want to separate a SIP calls...

If you are going with way 2, please implement such a REST endpoint. http://mycompany.com/wcs/connect and let us check

Hello 1. You can pass account details directly to JavaScript https://demo.flashphoner.com/client2/examples/demo/sip/phone/phone.html https://demo.flashphoner.com/client2/examples/demo/sip/phone/phone.js - SIP username - SIP authname - SIP password - SIP domain - SIP proxy - SIP port 2. You...

If most of RTSP streams you test are FullHD (1920x1080 etc), we recommend to keep this setting for better playback quality.

We confirm a 1-1.5 seconds delay in browser, raised the ticket WCS-4033, will let you know about progress. Please note: you do not provide SSH access to the server, so we can't use your server settings. We use the following parameters to reproduce the problem...

We checked the instance. The problem was in iptables settings: it is enabled by default, and it opens less ports than configured in inbound rules. We reset iptables iptables -F and WebRTC publishing and playback seems working now.

Try to switch server to TCP ice_tcp_transport=true Then restart server process. If it does not help, please provide SSH access via this form. We will check.

Дергание туда-сюда, это похоже на B-фреймы в потоке от OBS. Попробуйте выставить Max-B-frames в 0 на стороне OBS. B-фреймы в браузере не поддерживаются, пожтому играют только после транскодинга.

We answered in this topic.

Good day. Please check if IP addresses in ip and ip_local settings in /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties file are valid and equal to external and internal instance IPs respectively. You can also try to run the command cd /usr/local/FlashphonerWebCallServer/bin sudo...

To expand media ports range, change the settings in /usr/local/FlashphonerWebCallServer/conf/flashphoner.properties as we recommend above. Then configure the security group in the instance settings. Here is the default configuration You should change 30000-33000 to 20000-39999.

Seems like you need to expand media ports range: media_port_from=20000 media_port_to=39999 and RTSP agent ports range rtsp_port_from=40000 rtsp_port_to=50000

Добрый день. Мы протестировали звонки через Mango с предоставленными аккаунтами на последней сборке сервера 5.2.1895. С настройками сервера codecs_exclude_sip=mpeg4-generic,flv,mpv,opus,g729,speex16,g722 sip_force_tcp=true allow_outside_codecs=false звонки устанаваливаются корректно, обе...

You can send private information via this form 1. pcap 2. Two SIP credentials for a call through your PBX. We should be able to pass success Softphone-Softphone call, like Linphone - Linphone through your PBX.