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SIP issues with IP 406v2 (4.2.11)

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TheZCom

Programmer
Nov 2, 2005
122
US
I am currently running Broadvox SIP trunks to an IP 406v2 (4.2.11) behind a Sonicwall TZ 190. The Sonicwall has everything open necessary for SIP traffic. The pipe out to the internet is fiber-to-prem, so bandwidth is not an issue (we only have 3-4 calls max nailed up at any given time anyway). The calls are using G.711 codec. IPO has vintage VCM 10.

Calls sound great and we’re very happy with the setup/trunks for the most part. However, I have been experiencing the following and wondering where to start:

1. One-way audio occurs suddenly on an active call (“caller can hear us, but we can’t hear caller”). Of course I would understand if this happened here and there, but the issue occurs 10% of the time or so—much too frequently. I have noticed a pattern where this sometimes occurs when a user places a call when another user is already on a call (like getting ‘bumped’ off the call).
2. Call forwarding to outside number has no talk path whatsoever in either direction. For example, forwarding extension calls to mobile. Calls forward out, but no speech.

Any input regarding this scenario? Go to IP 500 with more recent VCM? Place a SIP gateway (Audiocodes type appliance) in between internet and IPO? Anybody have an IPO behind a firewall working really well with Broadvox SIP? Do you have call forwarding feature working? What would you do?

Your comments would be greatly appreciated!
 
Crap. It appears that the first test worked, but the call forwarded calls are back to the old one-way thing. **grumble**
 
You can't simply change the Firewall/NAT type. Well you can but it won't do anything other than tell the IP Office that you have one thing when that isn't indeed the case.

That value is best set by STUN discovery. If you run STUN and it comes back and says you have Blocking Firewall or Restricted Cone NAT then that is what you have. Trying to trick the IP Office into thinking that it is an Open Internet connection isn't going to accomplish anything.

If you want your one-way audio issue to go away you need it to come back with Open Internet or Full Cone NAT.

Kyle Holladay
ACA-I, ACA Call Center, ACS-I, ACS-M, TIA-CTP, MCP/MCTS Exchange 2007
ACE Implement: IP Office

"Thinking is the hardest work there is, which is the probable reason why so few engage in it." - Henry Ford
 
I've had to set the codec to G711Ulaw in the compression mode field on the SIP-trunk tab.
Otherwise I could only get one-way speech (tested on 4.2.11).
My laptop with 'phoner' a sip-client on the same internetconnection and with the same sip-account would auto-sense and switch to G711Alaw wich is more common in Europe I believe.
 
Update-I have the call forwarding feature working now. I set up LAN 2 for IPO outside the firewall (Sonicwall TZ 190). I will see if this addresses the one-way audio drop on concurrent calls. Thanks again for all of your suggestions! Z
 
Looks like that running sip via LAN 2 (outside Sonicwall), solved the reported issues in this thread. Once I set up the LAN2 items, I built a route to the provider to use LAN2. Seems to be working quite well now. Z
 
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