We cannot provide any support outside of WCS instance scope. You can request server setup, but we cannot configure third party VPN/firewall/etc which is outside of WCS instance.Can you guys provide any paid support for us ?? Please let us know .
Yes, it is necessary to import SSL certificates for WebRTC to work in browsers.1. Is import of ssl certificates is necessary in dashboard,is that the reason for blocking our media stream to server ??
You should get SSL certificate for servers domain name using AWS Certificate Manager for example. Then you should import SSL certificates as described here.2. If ssl certificates are necessary tell us how to get them with necessary steps ??
3.Our instance is launched in aws
Yes? you should import SSL certificates for your server domain1.Is it mandatory to install ssl certificate or we need to upload any certificate for playing the streams from our browser.
No, AWS Marketplace AMI already has a license billed hourly by Amazon2.Is license should be activated seperately in flashphoner dashboard(wcs server in aws market place)
Please deploy a new instance using this step by step manual. It should work out-of the-box. If this new instance does not work, provide SSH access to it using this form.3.From our end we are establishing connection successfully but publishing the stream from our laptop is not successful and we are disabling VPN server while streaming and disabled our security groups from our system level.
4.In aws we deployed the server as per documentation of flashphoner and ports are opened ,does we need to open other ports also for streaming?
Like you mentioned we opened the ports (31000-32000)/udp ports on either side but still streaming is not done?
No, you can publish and play WebRTC streams from desktop browser even with default self signed SSL certificates..5.At our dashboard level flashphoner is showing not secure in browser side ,is there any blocking occurs from browser level?
Yes, you can configure coturn on the same instance by this step by step manual.6.If nothing works can we go for coturn server in order to bypass the security group /NAT/firewall is it suggestable ?
You can get SSL certificate from AWS Certificate Manager or from Let'sEncrypt (use Centros docs because Amazon Linux 2 is based on Cemtos 7).1.If it is necessary to import ssl certificates where i can find this because we deployed server at aws and how can we get ssl certificate ??
The issue is in ports blocking, not due to SSL certificate2.As you were saying we have blocking something from our vpn/security groups side or it is due to ssl certificate ??
You can test with default self-signed certificates (don't forget to accept security exception in browser). Then, you can register domain name, get SSL certificate for this domain and import it using WCS dashboard.1.In the flashphoner dashboard also do we need to upload ssl certificate for getting the streams ?
2.ssl certificate is not mentioned while launching wcs server in aws ,do we need from server level to upload certificate ?
No, there is no any additional charge. Only the server instance itself is billed hourly by AWS.1. We have purchased the flashphoner server from aws market place and we are streaming up to 50 ip cmaeras ,So is there any additional charge for that ,pls let me know?
This depends on server capacity. Please read this article How many RTSP cams can I connect to the WebRTC server?2. How many ip cameras we can stream ?Is there any additional charge for this?
Please read this doc about HLS playback. Thre are 3 HLS samples:How can i play direct http protocol in flashphoner ? Is there any player in flashphoner dashboard?
Hello, I provided my SSH credentials for my server using this link; can you check our system? We have similar issue. Firewall ports are open:Provide SSH access to your WCS instance using this form, we will check.
Please create your own topic next time.Issue is when we publish a SIP as RTMP there is no audio. This has worked fine for a long time based on custom setup Flashphoner provided.
constraints: {
audio: true,
video: false
}
sipDomain
parameter ends with point sipDomain=pbx1.*******.**.
, but call is establishing normally.. 2021-09-29 09:54:39.545 Connecting to ***.**.***.*** port 222
. 2021-09-29 09:55:00.581 Failed to connect to ***.**.***.***: Network error: Connection timed out
/rest-api/call/terminate
. The hold music plays in softphoneadd_wcs_to_conference.sh
script to the extension number 7001, send DTMFIf you've updated Asterisk, this may be issue. Please review the PBX config, or roll back Asterisk update.The PBX is running ASterisk 18.6.0; I wonder if this is the issue?