[SteveW]

I am trying to setup an SPA3102 behind a Sonicwall TZ170 (Standard OS) using a Sipphone account. I am in the UK on an NTL cable network.

I have tried using the SIP transformation capability of the Sonicwall as follows:-

1) Sonicwall VoIP setting
Enable Consistent NAT - checked
Enable SIP Transformations - checked

SPA3102
SIP NAT Support settings
All set to no
Line 1 NAT Settings
NAT mapping enable - NO
NAT Keep Alive Enable - NO

When I try to connect to the Line one account ( which shows registered on the Info page) from a Gizmo project account on the same LAN. I get a ringing tone on Gizmo but nothing on the SPA3102. If I lift the handset I hear a dialling tone but do not connect to the incoming call. The 3102 info page shows the line 1 last caller, correctly, as the Gizmo account.

When I try calling back the Gizmo account I can connect but only have voice from the SPA3102 to the Gizmo account.


Next I tried using the Sipphone STUN server

2) Sonicwall VoIP setting
Enable Consistent NAT - checked
Enable SIP Transformations - unchecked

SPA3102
SIP NAT Support settings
Handle VIA received: Yes Handle VIA rport: Yes
Insert VIA received: Yes Insert VIA rport: YEs
Substitute VIA Addr: Yes Send Resp To Src Port:Yes
STUN Enable: Yes STUN Test Enable: Yes
STUN Server: stun01.sipphone.com:3478

Line 1 NAT Settings
NAT mapping enable - Yes
NAT Keep Alive Enable - Yes
Firewall policy Access rules:-
SIP port 5060 - Open to SPA3102 private address
RTP port 5004 - open to SPA3102 private address
RTP port min to max - Open to SPA3102 private address


As with 1) I can call the SPA3102 fromt Gizmo but get no ringing, but last caller is correct, and calling from the SPA3102 I can connect but only have voice from the 3120 to Gizmo.

Am I correct in thinking that Sonicwalls SIP transformations should handle all the NAT transformations or do I need to use any of the SPA3102's NAT support? As I get a similar error when I use 2) do I have another configuration problem. Also am I doing the right thing, trying to test the SPA3102 from a Gizmo account running on the same network?


Many thanks

[hwittenb]

Originally Posted by SteveW

am I doing the right thing, trying to test the SPA3102 from a Gizmo account running on the same network?

Could be fighting two problems at the same time unless you know the Gizmo softphone works OK. I would prefer to test by dialing a pstn number that answers with a recording, requires dtmf input, and has a way to record your voice. If that works, then worry about testing incoming calls.

The SonicWall looks pretty heavy duty, although the admin guide seems to have better documentation than many. I think you are correct that Enable Consistent NAT should always be checked. I would have thought the Enable Sip Transformations would have worked. The way I read it, it is supposed to look at the INVITE requests and open the correct rtp ports. The Gizmo phone, though, is using a non standard port for sip signalling. Maybe that is a problem.

[SteveW]

After checking a Sonciwall PDF document called Configuring VoIP for SonicOS Standard, have gone back to my first setup, allowing the Sonicwall to handle SIP transformations.

I can connect to the Gizmo echo test satisfactorally. Calling using Gizmo Projects does not reach the 3102 and after several rings I am transferred to voicemail. I have also tried a Call in number assigned to my account which transfers immediately to voicemail. The Sonicwall log shows packet dropped on UDP port 14868. Is this a SIP port?

Steve

[hwittenb]

Originally Posted by SteveW

The Sonicwall log shows packet dropped on UDP port 14868. Is this a SIP port?

A sip call uses ports for sip signalling and uses ports to exchange rtp (voice) packets. There will be a set of incoming ports (used by the 3102) and a set of outgoing ports used by your service provider.

For incoming Sip signalling the 3102 will use the port setup on the line tab. For incoming rtp packets the 3102 will use one of the ports in the range setup on the sip tab.

For outgoing transmission, the 3102 will send sip signalling to your provider's proxy at port 5060. Your provider will return (in the INVITE response) his ata port that you are to send the ata packets to.

In looking at the packets from the Gizmo phone, it was sending Sip signalling packet to the proxy server port 5060 and receiving them on port 64064. The Gizmo phone was receiving rtp packets using port 5004.

As for port 14868, Sonicwall could be changing the ports along the line. I don't know.