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  #1 (permalink)  
Old August 10th, 2008, 01:11 AM
root root is offline
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Default one way audio problem with IX67

Hi experts,

I'm using the internal SIP server of the IX67 and I'm trying to talk to a grandstream client (which is registered with my IX) over the internet. The GS client is behind a NAT and its IP address is 192.168.1.4.

My network configuration is as follows:
Phone1--[IX67]---[internet]---[NAT router]-{GS-486]--Phone2

- When I make a call from phone1 (2222) to phone2 (9999), phone2 does not even ring.

- When I make a call from phone2 to phone1, phone1 rings but I have one way audio problems (phone1 can hear phone2 but phone2 cannot hear 1)

It looks like the NAT is causing a problem. On the grandstream I tried setting mydomain.homeip.net as the proxy server but it didn't help.

On the IX67 I see phone2 registered with its local IP which is 192.168.1.4, is this normal?

I see the following the SIP log of the IX67. Btw, this network configuration works fine when I use the FWD SIP server/accounts instead of IX67's own server.

0d 00:18:36 t4 info send sf (0x8964b58) 127.0.0.1: INVITE sip:9999@mydomain.homeip.net SIP/2.0
0d 00:18:36 tSip info recv 127.0.0.1: INVITE sip:9999@mydomain.homeip.net SIP/2.0
0d 00:18:36 t1 info send sf (0x8db7c28) 192.168.1.4: INVITE sip:9999@192.168.1.4 SIP/2.0
0d 00:18:36 t4 info send sf (0x896693c) 127.0.0.1: SIP/2.0 100 Trying
0d 00:18:36 tSip info recv 127.0.0.1: SIP/2.0 100 Trying
0d 00:18:37 t1 info resend sf (0x8db7c28) 192.168.1.4: INVITE sip:9999@192.168.1.4 SIP/2.0
0d 00:18:39 t1 info resend sf (0x8db7c28) 192.168.1.4: INVITE sip:9999@192.168.1.4 SIP/2.0
0d 00:18:43 t1 info resend sf (0x8db7c28) 192.168.1.4: INVITE sip:9999@192.168.1.4 SIP/2.0
0d 00:18:51 t1 info resend sf (0x8db7c28) 192.168.1.4: INVITE sip:9999@192.168.1.4 SIP/2.0
0d 00:19:07 tSip error No answer from destination 192.168.1.4:5060

Any comments??

thanks in advance.

Boris
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Old November 14th, 2008, 02:36 AM
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herman herman is offline
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herman
Default Re: one way audio problem with IX67

Have you configured Phone 2 towards a STUN server. On that phone you would need the help from STUN and you would need NAT-traversal traffic. Alternativly you could make port forwarding in the NAT router for the used ports (SIP and RTP)
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