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You left only g729 codec for SIP calls, this codec does not supported by Chrome browser, so there's no sound. Choose another audio codec: opus, alaw, ulaw. Then comment the setting sip_force_tcp=true to disable TCP SIP support. Also, set this parameter dtmf=INFO and check if DTMF work with Sippy.

Hello. Please download latest Andoid SDK 1.0 build from this page if you plan to support Android 8 and earlier in your project, or Android SDK 1.1 build from this page, if you plan to support Android 9 or later. Then build Android examples as described on this page for Android SDK 1.0 or here...

A common recommendation for EC2 instances is to select an instance with 32 Gb RAM and 10Gb network, for example c4.8xlarge

To estimate resource requirements, please clarify the following: 1. How much streams published you need? 2. How much subsribers?

Hello. Stream snaphot taking JavaScript example is described here and on this page in details. Please check if snapshot is taken correctly with this example on your server. If Stream Snapshot example is working, please provide us an example with minimum code chenges that reproduces the problem...

h323 is not compatible SIP and H.264 is compatible and should be enough for call. Please provide a SIP account and room number. Then we will be able to make a call and test with DX80. support@flashphoner.com

Hello. The server.properties file does not used any more in WCS 5.1 and 5.2, please see this doc. You should set wss.port=443 to flashphoner.properties file.

In example above, we enabled G729 codec only. Try to enable more audio codecs. Also, if you do not use video calls at all, you can disable vp8 codec. Also, you can enable TCP usage for SIP signaling with this setting sip_force_tcp=true This eliminates SDP fragmentation, so no codecs exclusion is...

Hello. Unfortunately no, a user action is required to enable sound on autoplay in Chrome. Moreover, Firefox announces the same autoplay policy change in near future. In WCS 5.0 and 5.1, the WCS manager module performed web interface functions and stored admin password in its own database. In...

Сервер demo.flashphoner.com работает.

Please describe how the server is used and send the logs and configuration files to support@flashphoner.com.

WCS update from 5.1 to 5.2 is described here.

Hello. Please update to the latest build from this page and check again. If the issue persists, please prepare a report as described here, including gc-core-* logs and send to support@flashphoner.com, we will check. Also check what top command shows. Anyway, 2.91 Gb for 5 days does not look...

Hello. We fixed authentication issue in build 5.2.39. Please update and check. If problem perststs, please provide access to RTSP camera or prepare report as described here and send to support@flashphoner.com. Make shure you have captured RTSP packets to traffic dump.

Hello. We'll make a fix that should force DTMF sending even if no telephone-event codec in SDP (internal ticket WCS-1860). Also it seems like the problem is in WCS codecs configuration. In flashphoner.properties you have sent default codecs settings are used...

Hello. We work on it (internal ticket WCS-1861) and let you know when we fix {startTime} and add {endTime}

Hello. You can rename next recording part based on last modification time of previuos part. We'll try to reproduce it. Commonly, WCS should not toch a recording file after stream is actually finished. But, for example, pulled RTSP stream finished in 60 seconds after last subscriber is off, and...

Hello. We have tested the latest build 5.2.36 with OpenSIPS and Asterisk, with WCS default settings. WCS sends RFC2833 DTMF correctly, see traffic dump analisys screenshot So please update to this build and provide us traffic dumps for all of your PBXes where WCS does not work: Also, provide...

Your SIP gateway (PBX) answers in SDP SIP/2.0 200 OK m=audio 8376 RTP/AVP 18 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=sendrecv As you can see, your PBX does not indicate telephone-event (DTMF). And therefore your PBX does not work properly. Not WCS. You claim WCS 5.0 sends DTMF properly...

Hello WCS supports two DTMF modes: DTMF over RFC2833 DTMF over SIP INFO Your PBX does not support these modes. Third mode "DTMF Inband" is currently unsupported in WCS server. So to get this working you have to either migrate to another PBX or setup DTMF over RFC2833 / SIP INFO on your PBX...