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Author Topic: One Way Audio  (Read 17792 times)

Offline ralph

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One Way Audio
« on: August 26, 2011, 11:49:11 AM »
I'm in a bit of trouble.
I was brought in on a problem where inbound SIP calls ring in ok, the outside caller hears the inside person but the inside person hears nothing.
In talking with the firewall tech, he says the external IP address is NATing (PATing) 5060 to the 3300.
My question is this: When a call comes into the 3300, the 3300 sets up the call to the phone.
Once the user answers the call the 3300 drops out.   So how would the incoming audio hit the phones if FW policies only NAT to the 3300?

Ralph


Offline ccboe

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Re: One Way Audio
« Reply #1 on: August 26, 2011, 01:22:30 PM »
just thinking out loud

I would think you need to setup your sip device to use the 3300 as the proxy.

Tom

Offline brantn

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Re: One Way Audio
« Reply #2 on: August 26, 2011, 01:38:42 PM »
What kind of firewall? This issue is more then likely audio ports not being allowed through or SIP settings not being enabled in the firewall. SIP is only call setup so yes 5060 might be allowed through but what about the 5000 audio ports that can be used?

Offline ralph

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Re: One Way Audio
« Reply #3 on: August 26, 2011, 02:30:51 PM »
SIP Trunks - not devices
Juniper Firewall.

I'm part way fixed now.  - or at lease I was ~.5 hr ago -
Firewall tech reset the NATing and we started geting audio again.
Now we just cant transfer and get audo.
If we put the call on hold/off hold we loose audio.

Ralph

Offline brantn

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Re: One Way Audio
« Reply #4 on: August 26, 2011, 06:41:15 PM »
Try changing avoid signalling hold to Peer and let me know.

Offline ralph

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Re: One Way Audio
« Reply #5 on: August 26, 2011, 11:21:16 PM »
Tried that.  Didn't work.
I hate being in a position where I don't control of the firewall.  Too much finger pointing going on.
Two full days so far and it's still not right.

Ralph

Offline brantn

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Re: One Way Audio
« Reply #6 on: August 28, 2011, 12:15:25 PM »
Send your SIP Peer Profile to me just to verify also what is your Simulated CCM before ISDN progress in the trunks COS. After I verify those then I will blame the firewall.

Offline ralph

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Re: One Way Audio
« Reply #7 on: August 29, 2011, 09:15:42 AM »
Thanks brantn,
I've attached the SIP Peer Profile.
I checked the COS of the trunks for simulated CCM before ISDN Progress.  It was set to no so I changed it to yes.  No difference.
It's now back to no.

My current biggest problem is that if you put a call on hold or transfer, you loose audio.

Ralph
« Last Edit: August 29, 2011, 09:24:30 AM by ralph »

Offline bobcheese

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Re: One Way Audio
« Reply #8 on: August 29, 2011, 10:10:11 AM »
Hi Ralph, what version of software are you running on this 3300?

Offline ralph

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Re: One Way Audio
« Reply #9 on: August 29, 2011, 10:11:47 AM »
10.2.2.10

Offline brantn

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Re: One Way Audio
« Reply #10 on: August 29, 2011, 11:29:53 AM »
Avoid Signaling Hold to the Peer Set to Yes
Enable Mitel Proprietary SDP Set to No

Offline ralph

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Re: One Way Audio
« Reply #11 on: August 29, 2011, 11:36:09 AM »
Thanks brantn,
Made the changes but it made no difference.
I"m now looking at wireshark traces external to the firewall.
Tech support saw me sending invites to the carrier but got no responses.
Could be carrier.   Could be firewall.
Working through the trace now.

Ralph

Offline brantn

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Re: One Way Audio
« Reply #12 on: August 29, 2011, 11:52:30 AM »
With those 2 options set that should be a working config I am assuming then it is the firewall. I would look closely at the media ports as that is the typical issue is it is there just going to the wrong port because of consistent nat issues.

Offline ralph

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Re: One Way Audio
« Reply #13 on: August 29, 2011, 07:05:30 PM »
Firewall!
Tech reset the NAT policies and BOOM!  Audio!!

Still working though some other issues but at least we're up after 3 business days.

Ralph

Offline brantn

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Re: One Way Audio
« Reply #14 on: August 30, 2011, 04:19:05 AM »
Just a word of advice to all those sip lovers out there save yourself a lot of time and effort and use the ingate product line for sip termination it makes things a lot easier as well as that is what the majority of Mitel testing is done with.


 

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