https//:localhost:8444/ not opening

[Max]

Good day.
We've tested a call from outside network through your WCS (from Chrome Browser to softphone). It works for us, both sides are hearable for each other. Please note that local SIP PBX address should be set as SIP domain and SIP proxy when you're registering in browser:

 

[SAGARVG]

Good day.
We've tested a call from outside network through your WCS (from Chrome Browser to softphone). It works for us, both sides are hearable for each other. Please note that local SIP PBX address should be set as SIP domain and SIP proxy when you're registering in browser:
View attachment 2540

we tested this this working absolutly fine with local IPPBX...but if i check this same with cloud virtual IP PBX, i am able to get registered but not able to exchange the voice

whether we can do the video streaming with this call, and receive video call as well, i checked with phone video of SIP not able to make .....video exchange via call

form updated for SIP numbers and SSH

 

[Max]

Hello

Server 192.168.10.12 must be located in DMZ zone.
At least TCP UDP ports in range [30000-32000] for SIP, voice, video traffic must be reachable.

For example, your host has public static IP address 9.9.9.9
In such a case if UDP or TCP packet received on 9.9.9.9 port i.e. 31500, then host 192.168.10.12 should receive this packet on port 31500.

Please check port mapping using nc.

Accessory tools - Web Call Server 5.2 - EN - Flashphoner Documentation

docs.flashphoner.com

Example:

1. Launch tcpdump on host 192.168.10.12
tcpdump udp port 31500

2. Do check 31500 port from outside host
echo -n "hello" | nc -u -w1 9.9.9.9 31500

Here 9.9.9.9 is public static IP of your LAN host.

See screenshot. We checked port 31500 and port is unreachable.

 

[SAGARVG]

ample, your host has public static IP address 9.9.9.9

TCP UDP ports are port forwarded 31000-32000 checked the packet as mentioned in screenshot

but issue remain same ,voice exchange or video exchange not happening, but able to get registered

 

[Max]

Please test the call again and gather the traffic dump

tcpdump -i eno8 udp -s 4096 -w log.pcap

If you open this dump in Wireshark you should be able to see voice traffic.

Send us this dump for analyzing either via form or directly into the forum thread.

Note. The test call should be started AFTER starting the tcpdump session.

 

[SAGARVG]

Please test the call again and gather the traffic dump

tcpdump -i eno8 udp -s 4096 -w log.pcap

If you open this dump in Wireshark you should be able to see voice traffic.

Send us this dump for analyzing either via form or directly into the forum thread.

Note. The test call should be started AFTER starting the tcpdump session.

form has been updated

any SSL certificate i must import???

Please check

 

[Max]

Unfortunately, SIP traffic cannot be identified in pcap dump. Perhaps you've started dump collection after call establishing, but it must be started before this.
On your server we saw the following messages in server logs:

13:13:08,673 WARN    AbstractStunSocket - STUN-UDP-pool-40-thread-1 Can not find local candidate for /192.168.10.20

So the following setting should help:

rtc_ice_add_local_interface=true

 

[SAGARVG]

Unfortunately, SIP traffic cannot be identified in pcap dump. Perhaps you've started dump collection after call establishing, but it must be started before this.
On your server we saw the following messages in server logs:

13:13:08,673 WARN    AbstractStunSocket - STUN-UDP-pool-40-thread-1 Can not find local candidate for /192.168.10.20

So the following setting should help:

rtc_ice_add_local_interface=true

we are testing webcall server to webcall server its not exchanging voice and video

but with IP intercom with webcall server its working and vice versa its working

 

[Max]

i am not able to access webcall server outside network but i was accessing before...

We also cannot access the server by SSH credentials you've sent

Seems like something broken on router/NAT, please check.

 

[SAGARVG]

We also cannot access the server by SSH credentials you've sent
View attachment 2656
Seems like something broken on router/NAT, please check.

we are testing webcall server to webcall server its not exchanging voice and video

but with IP intercom with webcall server its working and vice versa its working
remote access of webcall server is ok after firewall restart

form has been updated,

 

[Max]

Seems like you're opening the example page locally from disk file:

18:06:51,664 INFO       WSServerHandler - WSS-pool-23-thread-19 Orgign: file://

Please make sure you open the page via HTTPS, for WebRTC to work.
Also please note that SIP PBX is probably misconfigured. We cannot establish a call between two softphones, or from softphone to browser, or vice versa using accounts you've provided:
1. If both account (A and B) are registered, call fails, SIP PBX sends 487 Request Terminated
2. If only one account (A) is registered, call to another account (A calls B) is establishing, we can hear long beeps (and, if using your WCS, can hear trial license watermark).
So we recommend SIP PBX setup too.

 

[SAGARVG]

Seems like you're opening the example page locally from disk file:

18:06:51,664 INFO       WSServerHandler - WSS-pool-23-thread-19 Orgign: file://

Please make sure you open the page via HTTPS, for WebRTC to work.
Also please note that SIP PBX is probably misconfigured. We cannot establish a call between two softphones, or from softphone to browser, or vice versa using accounts you've provided:
1. If both account (A and B) are registered, call fails, SIP PBX sends 487 Request Terminated
2. If only one account (A) is registered, call to another account (A calls B) is establishing, we can hear long beeps (and, if using your WCS, can hear trial license watermark).
So we recommend SIP PBX setup too.

BOTH 1001 AND 1005 are configured as ip phone

1001 and 1005 we can test for web call server ,where as 1001 to 1004 working fine both way

 

[Max]

We expiriencing the issue with all the sofphones used during the test (but all of them work with OpenSIPS and Asterisk). Please clarify what softphone do you use for test?
Anyway, WebRTC traffic flows correctly during our tests with your server. We opened Phone Min example page via HTTPS, as we recommended above.

 

[SAGARVG]

We expiriencing the issue with all the sofphones used during the test (but all of them work with OpenSIPS and Asterisk). Please clarify what softphone do you use for test?
Anyway, WebRTC traffic flows correctly during our tests with your server. We opened Phone Min example page via HTTPS, as we recommended above.

we are opening video call from Dashboard in that case i am getting error like this ,where as in webpage its working

 

[Max]

Ths seems like some error in Javascript code. Please add more logging in your code (using console.log() function for example) to check if variable call is defined in onclick event handler.
Or you can provide Javascript source code you test using this private form, we will check. The code should be minimal to reproduce the issue, no PHP, no frameworks.

 

[SAGARVG]

Ths seems like some error in Javascript code. Please add more logging in your code (using console.log() function for example) to check if variable call is defined in onclick event handler.
Or you can provide Javascript source code you test using this private form, we will check. The code should be minimal to reproduce the issue, no PHP, no frameworks.

form has been updated

 

[Max]

Please test HTML+JS example just like here. No proprietary JSON which we cannot even run to test, no frameworks, just a plain code. If this doesn't work, provide the code using this form. Or provide the link to your test page, we will test in browser directly.

 

[SAGARVG]

Ths seems like some error in Javascript code. Please add more logging in your code (using console.log() function for example) to check if variable call is defined in onclick event handler.
Or you can provide Javascript source code you test using this private form, we will check. The code should be minimal to reproduce the issue, no PHP, no frameworks.

test page link updated in that u can see voice, if need to edit then widget library voice

 

[Max]

Please provide a detailed step by step instruction how do you test. What exactly should be done to reproduce the problem?

Anyway, the link is HTTP. WebRTC will not work via HTTP in browser. Please use HTTPS instead.

 

[SAGARVG]

Please provide a detailed step by step instruction how do you test. What exactly should be done to reproduce the problem?
View attachment 2679
Anyway, the link is HTTP. WebRTC will not work via HTTP in browser. Please use HTTPS instead.

form has been updated about opening file