Max's latest activity

You can change the buffer size settings: rtmp_out_buffer_enabled=true rtmp_out_buffer_polling_time=20 rtmp_out_buffer_start_size=10000...

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...

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...

Good day. Please clarify the case: 1. Caller establishes audio only call from browser to the TTS 2. Caller hears audio from the TTS...

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...

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...

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...

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...

Please collect a report as described here including a traffic dump and send using this form.

We reproduced the issue and raised the ticket WCS-4735 to fix. Will let you know about progress.

Use rtc_ice_add_local_interface=true instead. In this case, ip_local should be added to candidates list.