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]

I have a Digisol if you have seen the screenshots. Nothing special in firmware to make it work or not work, just configure properly and things will work.

 

[AARON NASH]

Hi @sloj, I tried to configure my cousin's Netlink 323rgw ONT(GPON ONT ONU WITH TELEPHONE WIFI INTERNET LAN FTTH) with firmware version 1.0.29, Zoiper is registered but only one-way audio is supported. Then I updated the ONT's firmware to 1.0.35 and now Zoiper5 is working fine. And also the old firmware was getting stuck after adding a static route.

You can also try to update your Digisol Ont's firmware to 1.0.35. All these ONT brands like Netlink, Digisol, Syrotech, DBC are importing from Chinese manufacturer/Company V-SOL(V-SOL HG323RW 1GE 1FE 1FXS GPON WIFI ONU) and later rebranding and distributing these ONTs in India. They all have the same hardware and same web interface/firmware.

 

[sloj]

Ya some versions do have bugs that stop it from working properly. I use 1.0.34 and it works fine.

 

[bigbyte.me]

Thanks to excellent information available on this forum, specially by @sloj and @varkey , I was able to test successfully voip on microsip and gswave lite on Android.
My hardware is Syrotech WiFi ONT 1110-wdont, hardware version 2.0 software V2.0.5-24566.
I tested Internet in bridge mode though I didn’t test voip with a bridged router.
Strangely, I initially faced no audio issue in microsip on windows ( ip rewrite enabled though) after first successful call. Enabling SIP ALG in ONT solved this issue and later same configuration worked successfully for Android as well.

Also my ISP is MTNL Mumbai but I am posting here as I learnt all information from BSNL subsection of this forum.

Next, I am planning to flash my R7000 with Freshtomato and test dual wan and vlan binding after bridging both voice and data as shown by @varkey in other thread.

Once again I’d thank all the members of this forum for sharing their findings and helping other members here. Long live Internet.

 

[bigbyte.me]

@sloj and @varkey @AARON NASH
During the weekend I played with some configuration and tried some scenario suggested by members in the group, so here is the update and some findings.
I tried to source a unused router from friends where I can install openwrt, Got TP-Link 841N V11. Installed open wrt on it. Now this router has separate Nic for WAN so it doesn't appears in switch configuration. Had to reassign a LAN port as WAN.

Made interfaces in open-wrt with proper VLAN tagging. Configured ONT with VLAN binding at port 2, Failed Couldn't connect.

Puzzled me, removed the VLAN binding at decided to test just internet. So removed VLAN Binding from port 2 and activated port binding of port 2 for internet wan configuration in ONT. Failed again. (However, in this configuration, if from router side I make untagged normal PPPOE it works, as tested earlier).

Willing to do more experiment, I changed VLAN mode in ONT from Tagged to Transparent. Voila got connected from router.

Somehow my ONT is not binding VLAN's from Router and ONT properly, or may be I am doing something wrong.

TL;DR, If interface with VLAN id is configured in both ONT and router I am unable to establish connection, both in Port binding or Vlan binding mode of ONT port.

 

[Siddharth070999]

@sloj can u dm me?or anydesk?

 

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[Siddharth070999]

i am unable to ping my sip server @sloj..plzz help if u r available.

 

[sloj]

Follow the guides. Frankly, I am tired.

 

[Siddharth070999]

i tried to ping sip server but it said request timed out..anyways

 

[spantktm]

@sloj Please help. Using Sharp AS 341 WT router (interface is exactly similar to your screenshots) . After adding route I am able to use soft SIP clients. Incoming and Outgoing calls are coming but there is no voice on either side.
Things I would like to tell u :
1. My voice profile as configured by LCO is in DHCP with 1830 vlan. (phone no starts with 297, took new FTTH connection in 2020 around september; Its not converted no like from landline to broadband and similar)
2. Voice also works in Static apart from DHCP (I copied IPs and gateways from status page while on DHCP).
3. On landline I am able to use calls perfectly, without any problem
4. My SIP IP is a lot different from what many people (10.x.x.x.) post here, mine is 192.168.149.78. So I have added a route starting from 192.x.x.x. . If I use 10.x.x.x. SIP client do not connect.
5. As per some youtuber's suggestion I changed my router IP to 172.168.1.1 from 192.168.1.1 (He said your SIP IP starts from 192.x.x.x.) , so change router IP otherwise they might clash.
6. As per your guide I am using microsip and Allow IP rewrite is enabled.

Calls and coming and going. Only problem is there is no Audio.

I don't really know a lot about these things. So, please if you can help via Anydesk. ( I am unable to DM you as I am a new user on this forum)

 

[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.