https//:localhost:8444/ not opening

[SAGARVG]

We still cannot access your server with credentials you've provided, as we mentioned above. Please check.

private form updated ,can you please check with SSH

 

[SAGARVG]

We still cannot access your server with credentials you've provided, as we mentioned above. Please check.

SSH private updated ,can you please check now, I am not able to register SIP phone in the webcallserver only i am getting message established

 

[Max]

We tried to make a test call via your server again, with the same result:

The ports 8443, 8444 seems to be opened locally on server, but are closed on router/NAT.
Please provide a public instance (AWS, DO, Google Cloud, Yandex Cloud etc) and a public SIP server to test, or provide a fully working access from outside to your private network. Otherwise, we cannot make a test call via your server, so we cannot locate the problem and help you to solve it.

 

[SAGARVG]

We tried to make a test call via your server again, with the same result:
View attachment 2350
The ports 8443, 8444 seems to be opened locally on server, but are closed on router/NAT.
Please provide a public instance (AWS, DO, Google Cloud, Yandex Cloud etc) and a public SIP server to test, or provide a fully working access from outside to your private network. Otherwise, we cannot make a test call via your server, so we cannot locate the problem and help you to solve it.

8444 and 8443 are port forwarded can you please check

SIP server related ports are forwarded can you please check 5061

public instance 8089 for cloud

 

[Max]

We finally checked your server.
Seems like you've set a wrong IP address in ip and ip_local settings

So WCS cannot bind SIP signaling port at all.
When changing IP address settings, WCS binds SIP signaling port (30001 for first SIP registration by default) and sends SIP REGISTER to SIP server. But the SIP server listens neither 5060 nor 5061 UDP ports:

So to fix the issue you should:
1. Set right IP address to WCS configuration file flashphoner.properties then restart WCS
2. Configure SIP server to listen UDP SIP signaling port (5060, 5061 or what you're using)
3. Optionally, switch SIP signaling to TCP in WCS configuration

sip_force_tcp=true

and set up SIP server to support TCP SIP signaling.

 

[SAGARVG]

I changed IP address in config file


SIP server already opened with 5061 /5060 port which we tested with Soft phone outside network


I can register but voice is not able to transfer from webpage to SIP Phone but voice can be transferred from SIP phone to Web page

 

[Max]

With SIP credentials you've provided we receive 403 Forbidden response from SIP server, but REGITER flow looks normal. Please test a call. If this still cannot be established, please provide actual SIP credentials using this form, we will check.

 

[SAGARVG]

With SIP credentials you've provided we receive 403 Forbidden response from SIP server, but REGITER flow looks normal. Please test a call. If this still cannot be established, please provide actual SIP credentials using this form, we will check.

form has been updated, one thing observed Transport TLS SIP number are not getting registered, where has UDP+TCP transport getting registered in SIP IP PBX,,what must be the issue

 

[Max]

We just tried to register with both accounts you've provided. The SIP server breaks the connection after first SIP REGISTER packet sent to port 5061:

but when we trying to register to the port 5060, the flow goes correctly (but server returns 403 Forbidden):

Unfortunately, we cannot check connection to your SIP server using third party softphone, so we cannot confirm the problem at WCS side.
Please configure TCP or UDP transport at SIP server side.

 

[SAGARVG]

We just tried to register with both accounts you've provided. The SIP server breaks the connection after first SIP REGISTER packet sent to port 5061:
View attachment 2357
but when we trying to register to the port 5060, the flow goes correctly (but server returns 403 Forbidden):
View attachment 2358
Unfortunately, we cannot check connection to your SIP server using third party softphone, so we cannot confirm the problem at WCS side.
Please configure TCP or UDP transport at SIP server side.

can you check with TCP _UDP transport form has been updated


One more thing i followed Embed file in this location

Embedding a browser-based SIP phone to a web page | Flashphoner Streaming & Calls for Web

flashphoner.com

After i call i get the following message without any call going or receiving, this will use in our dash board for calling purpose

 

[Max]

can you check with TCP _UDP transport form has been updated

We tested again. The latest accounts you've provided are successfully registered on SIP server, but call cannot be established because WebRTC media ports (31001-32000/udp by default) of WCS server are not available from outside:

Please check if the ports range is available from PC where you opening page to test using this method for example.

 

[SAGARVG]

We tested again. The latest accounts you've provided are successfully registered on SIP server, but call cannot be established because WebRTC media ports (31001-32000/udp by default) of WCS server are not available from outside:
View attachment 2361
Please check if the ports range is available from PC where you opening page to test using this method for example.

ports 31000 to 32000 are forwarded same verified can you please check,
i checked by using packet sender tool where i sent using port 31006 ,and same reflecting in firewall is forwarded to webcall server

attached image of packet sender and firewall

 

[Max]

We've checked your environment.
To make a call from outside of your LAN, you should change the following in WCS configuration:
1. Set ip parameter to external IP:

ip=external_ip
ip_local=internal_ip

2. Add the following parameters

client_mode=false
rtc_ice_add_local_component=true

In this case, a SIP call between accounts you've provided is made successfully using Phone Min example in Chrome browser.

 

[SAGARVG]

We've checked your environment.
To make a call from outside of your LAN, you should change the following in WCS configuration:
1. Set ip parameter to external IP:

ip=external_ip
ip_local=internal_ip

2. Add the following parameters

client_mode=false
rtc_ice_add_local_component=true

View attachment 2369
In this case, a SIP call between accounts you've provided is made successfully using Phone Min example in Chrome browser.

after changing the configuration issue remain same

if i am calling inside network between webcall server and SIP phone it works fine...

If i am calling from outside network from SIP phone to webcall server works fine

But if i am calling from webcall server to SIP phone outside network its not working fine


i followed Embed file in this location

Embedding a browser-based SIP phone to a web page | Flashphoner Streaming & Calls for Web

flashphoner.com

After i call i get the following message without any call going or receiving, this will use in our dash board for calling purpose

 

[Max]

Please check if the issue is reproducing in Phone Min example.
If not, please modify Phone Min example code minimally to reproduce the issue and send using this form.
Also please check if SIP call establish successfully using Phone Min example on our demo server using public SIP PBX.

But if i am calling from webcall server to SIP phone outside network its not working fine

This seems to be NAT issue. To check this, please provide a public SIP server credentials to access from outside network to it using this form.

 

[SAGARVG]

Please check if the issue is reproducing in Phone Min example.
If not, please modify Phone Min example code minimally to reproduce the issue and send using this form.
Also please check if SIP call establish successfully using Phone Min example on our demo server using public SIP PBX.

This seems to be NAT issue. To check this, please provide a public SIP server credentials to access from outside network to it using this form.

phone min example. form has been updated .


in demo server i am not able to register the SIP phone

SIP credential provided in the form

 

[Max]

A SIP call is sussessfully established using your code example in two different browser windows

This is confirming by traffic dump (we marked RTP traffic going through the server)

So your code seems to be working.

in demo server i am not able to register the SIP phone

You should provide a public IP and port (not a private 192.168.*.*) to access your SIP PBX to register SIP phone from outside network.

 

[SAGARVG]

A SIP call is sussessfully established using your code example in two different browser windows
View attachment 2372
This is confirming by traffic dump (we marked RTP traffic going through the server)
View attachment 2373
So your code seems to be working.

You should provide a public IP and port (not a private 192.168.*.*) to access your SIP PBX to register SIP phone from outside network.

we used same html in dashboard ,but it creates error where as in browser works fine

not able to register only established

 

[Max]

we used same html in dashboard ,but it creates error where as in browser works fine

When opening your sample html code locally in Chrome browser, it works fine without errors in console (see this post).
We recommend you to remove this string from code because it seems excessive

<script type="text/javascript" src="https://github.com/flashphoner/flashphoner_client/blob/wcs_api/client/src/examples/min/phone/Phone-min.js"></script>

Seems like the problem is in your dashbord, for example, script is not fully dowloaded. We also recommend you to place flashphoner.js to your webserver to prevent donwnloading issues.

not able to register only established

Seems like your SIP PBX is unreachable from outside using the address and port: the TCP connection is accepted, but REGISTER request is not anwsered:

The same result both for registering from browser via WCS and from SIP sofphone (MicroSIP for example).

But if i am calling from webcall server to SIP phone outside network its not working fine

This could be the same issue.
Please check your router.

 

[SAGARVG]

When opening your sample html code locally in Chrome browser, it works fine without errors in console (see this post).
We recommend you to remove this string from code because it seems excessive

<script type="text/javascript" src="https://github.com/flashphoner/flashphoner_client/blob/wcs_api/client/src/examples/min/phone/Phone-min.js"></script>

Seems like the problem is in your dashbord, for example, script is not fully dowloaded. We also recommend you to place flashphoner.js to your webserver to prevent donwnloading issues.

Seems like your SIP PBX is unreachable from outside using the address and port: the TCP connection is accepted, but REGISTER request is not anwsered:
View attachment 2381
The same result both for registering from browser via WCS and from SIP sofphone (MicroSIP for example).

This could be the same issue.
Please check your router.

i can register the same using zoiper softphone outside network, but with webcall server-5 its not getting registered...

In dashboard i removed the mentioned lines and copied the flahphoner.js locally its working fine in the web browser , but in dashboard giving error as attached