[roberthuang]

I posted this topic on www.dslreports.com but couldn't get answer, I hope somebody here can help me.

I have a SPA 2002 and set VoipStunt on line 1 while Vbuzzer on line 2. Peer-To-Peer callings between vbuzzer users are working.However it won't work between voipstunt users. My settings are shown below(UDP 5060 and 3478 have already been forwarded to my Sipura on the router):

User ID: My VoipStunt username
Password: My VoipStunt password
Proxy : sip.voipstunt.com
Use Outbound Proxy: No
Register: Yes
STUN server: stun.voipstunt.com:3478

I can make international calls through VoipStunt without any problem. I put OtherVoipStuntUser@sip.voipstunt.com:5060 in Speed Dail 2, when I press 2#, Callee's phone rings, but no sounds on both sides.

Anybody succeed in making peer-to-peer callings between voipstunt users? I need your opinion. Thanks in advance.

[DracoFelis]

Originally Posted by roberthuang

User ID: My VoipStunt username
Password: My VoipStunt password
Proxy : sip.voipstunt.com
Use Outbound Proxy: No
Register: Yes
STUN server: stun.voipstunt.com:3478

I can make international calls through VoipStunt without any problem. I put OtherVoipStuntUser@sip.voipstunt.com:5060 in Speed Dail 2, when I press 2#, Callee's phone rings, but no sounds on both sides.

1) What are your other STUN settings (i.e. how are you using STUN)? Some STUN setups work better for SIP URI calls (not to mention P2P calls) than others. I'm personally pretty fond of the settings shown in the attached graphic.

2) Do you (and the other person) have your routers forward the SIP and RTP (voice) ports to you adapters? Some calls (especially "peer to peer" calls) are sensitive to this forwarding, even when "normal calls" will go through without it...

3) If all else fails, remember that while many providers allow inbound SIP URI calls, many others block such calls. If VoipStunt is one of the latter providers, you might want to consider bypassing them entirely. Here's one trick that allows for "true" P2P calls (bypassing all service providers):
http://faq.sipbroker.com/tiki-index....%20or%20Sipura

Originally Posted by roberthuang

Anybody succeed in making peer-to-peer callings between voipstunt users?

What you are currently attempting is NOT a true "peer to peer call". What you are attempting is a "SIP URI" call via VoipStunt. That will only work if VoipStunt allows such calls (and I don't know if they do, or they don't)!

For "the gory details" of a true "peer to peer" (direct between VoIP adapters, bypassing all service providers) call, follow the link I posted in point 3...

[roberthuang]

Originally Posted by DracoFelis

2) Do you (and the other person) have your routers forward the SIP and RTP (voice) ports to you adapters? Some calls (especially "peer to peer" calls) are sensitive to this forwarding, even when "normal calls" will go through without it...

Yes, RTP ports from 16384 to 16482 have also been forwarded to my SPA.

I have sent an email to VoipStunt regarding if they support SIP URI call or not. Now waiting for their reply. I will try another Stun server.

Thank you DracoFelis.

[jimng78]

Hi,

Knowing that all you guys are voipstunt users and I want to seek some info about the problems that encountered. I have an account with an incoming Swiss number and I am using Grandstream 486. The problems that I have that the ATA couldn't be registered with voipstunt since this morning, but it worked when using the softphone downloading. The 486 worked pretty good days before and I haven't changed any setting since then. Also, the incoming function doesn't work properly with the 486, I have to forward to my ceel to get incoming call but not from the phone connected with 486. I have tried to contact the customer service of voipstunt before but got no luck to get response from them.

Wish somebody can help in this aspect.

Jimmy