FTTH VOIP SIP Softphone configuration with ONT/ONU. (Now works on some more apps)

After hours of fiddling with my ONU and SIP apps, here is a guide on how to get it working.

First login to the ONU/ONT and change these things


Change VOICE to VOICE_INTERNET.


Add a Static Route to the SIP Server. You can add the IP directly like I did in my previous post. But that way you will have to add the return route too which is not there in my previous post. In my case, the return route is 10.x.x.x which is the same with my SIP server so I have to add only one route entry and be done.


If SIP is disabled, enable it and disable it again. This is necessary because SIP ALG functionality is broken. If it worked the way it was supposed to we could have easily setup softphones on our devices.


This is your VOIP config page. You wil find your SIP Server IP here. The proxy part isn't really necessary.

Also, you can disable IGMP on Voice. It's unnecessary.


Make sure NAT is enabled. Port Binding isn't necessary as we will be doing the setup on PC for now directly via ONT/ONU without any external router.

Hop to PC

Install YATE or MicroSIP and configure them with your SIP details. These are the two I have used that support NAT.

This goes without saying but just putting it out there. Allow the apps Windows Firewall prompt to connect to public network (or any other firewall you are using).

Configuration Screenshots


This is for MicroSIP. Click on the top right corner button -> Add Account. Don't forget to tick Allow IP Rewrite.


Another one for MicroSIP. Click on the top right corner button -> Settings. RTP ports range doesn't have to be that wide as I have in the Screenshot. 4000-4001 or 4010 is fine.


This is for YATE. Go to Settings -> Accounts -> Click on New


Resource Monitor showing MicroSIP info when it's idle not dialing any number.


Resource Monitor showing MicroSIP info when I dialed a number. See the change under Network Activity. Now there is another IP which will receive voice and reply which I will receive on my static VOIP IP which NAT will take care of and send to my local IP.

Link to previous failed attempt.

New changes that will let VOIP work on some more apps and platforms.

Tick "Turn off LAN DHCP" on Voice Internet configuration and Enable SIP ALG. Didn't think disabling DHCP makes SIP ALG function properly. Really hard to know which option does what without a proper manual.

Also, those who have DHCP instead of Static in Voice Internet configuration, everything else is same.

 

[sloj]

If there is no audio, chances are you have selected the wrong audio input and output in app or RTP port may be blocked. I am a little preoccupied with other things right now and can't come to Anydesk. Please go through the entire thread and see if you missed something.

That being said, I would advice not to use this as a permanent setup and use the phone service like a regular landline setup. It was just curiosity to see if it can be done which led to this.

 

[spantktm]

Thanks for such an early reply. Forgot to mention, that I have cross checked with audio input output devices and sound control panel. Also checked using a bluetooth earphone, still no luck.

Regarding RTP port, I don't know anything. I understand you are busy these days.

Still if you get any time of your convenience please ping me. I can wait for this. This will help me a lot. Actually I want my mom to easily use VoIP calls using her smartphone instead of landline.

Thank you again.

 

[sloj]

The reason I said not to use is because SIP apps drain battery a lot with UDP. I haven't checked if it can be used with TCP or not. I may have to test it. May be this weekend.

Also the issue is with incoming calls which eats more battery because the app needs to keep checking if the link is still active. There is a timeout period after which the app needs to register again to keep the connection alive and going which in turn contributes to battery drain.

 

[spantktm]

I understand what you are telling is for my good. But if this SIP voice problem is sorted out, it will save 400-500 rs of mobile recharge. My mom usually makes very less calls. So buying a recharge just for those, doesn't seems justified to us. Its just waste of money.

There I want it to be configured on her phone. After many efforts outgoing and incoming are working. But don't know what is the problem with audio. Have tried almost everything on internet.

That's the reason I am requesting you to have a look at it anytime you are free.

BTW if u could DM me, I will tell you the exact problems.

 

[sloj]

Get a cordless phone, problem solved.

You should post here so that if someone else is stuck in the same situation it might help them in the future. Will DM you later.

 

[spantktm]

Okay. will wait.

 

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[@Tx_JK]

Im actually stuck with the a similar problem.
I need to open up port 5060 to register SIP on native Android dialer.
To open it u have to disable SIP ALG and this will end up having no voice in both Zoiper and Android.
I'm not sure what to do. Port forwarding or changing NAT config to NAT3 or NAT2

 

[airtel_sip]

Hello, I really need help with SIP.
I don't have a landline, I need to make calls from PC or android phone.
I can't ping to ims.airtel.in
I can't understand the configuration because many things have changed on my router.
I have the details. Can someone help me on teamviewer or anydesk?
I will be so grateful. Thanks a lot!

 

[airtel_sip]

@sloj What do I need to do first? Here is my router page.

 

[nsniteshsonal37]

Can someone help me set this up with my Syrotech ONT? I can't find my gateway for the routing.