Remote 9608 Audio Problem 1

[phonesaz]

thread940-1681769 is probably the best thing I have read regarding remote teleworker phones on the IPO. Having said that, I am at an impasse on two different installations so am wondering if I am understanding the setup properly.

The simplest of the two installations uses a Linksys RV0041 router. I have done the following: Forwarded 1719, 1720, and the RTP range 49152-53247 and verified they are in the table of the router. It seems like to allow ICMP I needed to uncheck the box under the firewall settings that says "block WAN request" (description from the manual below), so I did.
From the router manual:

Block WAN Request: This feature is designed to prevent attacks through the Internet. When it is enabled, the
Router will drop both the unaccepted TCP request and ICMP packets from the WAN side. The hacker will not find
the Router by pinging the WAN IP address

I used a stun server that fed back the correct public IP address and Full Cone NAT as the firewall type. The phones are programmed correctly; I installed them on-site, built the user and extension, verified they could both make and receive calls, and then brought them to my home office, deleted all the settings, and put in the public IP address as the call server. The phones boot up, register, show the correct extension, but have no dial tone and/or audio path. I have de-selected direct media path, but my understanding is the system ignores it anyway? The phones show as registered under H323 status in monitor, I see the phone and can ping it from within system status, the phone has the proper name, extension, etc, but I have no audio. I can dial a number, my cell phone sees the incoming call and I can answer it, but no audio either way.

This is the second system I have had this happen with so I am starting to wonder if I am missing something in the setup. I don't think I am, because two others I have done work perfectly. The other one that doesn't work has a Watchguard firewall, but this one just has a basic router. They are able to use voice mail to email and have sip trunks working; I just don't know where to go to troubleshoot this audio issue. Any input would be GREATLY appreciated!! Thank you.

 

[holdmusic34]

You're setting up a remote extension in this set up, not a VPN connection.

In a proper VPN connection, the VPN handset will drop in to the network courtesy of the router and the IPO will have no ide it's a remote extension. You shouldn't need NAT.

No audio is your RTP stream not able to pass through packets.
To persist with this setup:
Tick remote extension enable.
Use the Port Number Range (NAT) setting instead of RTP port number range.
Set you ports to 55000-55010. This will give you 5 remote extensions.
Forward these ports to your IPO.

A VPN is much more preferred.
The RTP range you have selected encompasses many IP Office ports that are used for software such as monitor.


I'm clear but my fighter is down.

 

[phonesaz]

Yes, even though the original thread was about a 9608 VPN, most of it veered to the topic of remote H323 and that is what I am trying to implement. I am clueless as to why sometimes it works by following all the steps and sometimes it doesn't. I have the remote extension and remote gatekeeper enabled, and the users are marked as remote workers. Are you saying I should change the forwarded ports from the 49152-53246 to 55000-55010 or use both? I am going to add the 55000-55010 and see if it makes a difference.

I have been reading the system monitor document that came as an attachment to the R9.1 Avaya training class. It doesn't seem very specific on how to set traces/filters to determine where the issue might be in a situation like this. It seems like with the right filters I could figure it out. Thank you.

 

[phonesaz]

So.. I changed the ports on the NAT to the 55000-55010 and added that rule to the firewall, deleted the large RTP range, rebooted made no difference. Then I read some more threads and saw one from tlpeter that suggested disabling 801.1, which was set to auto. I changed it to off, rebooted, and now I have audio. Not sure which did it, or a combination of both, but I much prefer 10 ports open instead of the Avaya suggested range and did I mention it works? Thank you again for your help holdmusic34.

 

[holdmusic34]

Correct.

In a SIP trace you'll see an RTP Port stream as per below. Look for something similar in a H.323 trace.
Start by going Monitor> Filter> Trace options> Clear All. Select Print and error filters only and then add what you need from there.
Also check your system status licenses> have you got enough licenses for IP endpoints? You can only have four remote phones for free without an extra license.

1103633369mS SIP Reg/Opt Tx: phone
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.230:5060;rport;branch=z9hG4bK-1a8cac-67b583a2-41590ee8
From: <sip:795@192.168.8.15>;tag=4ba6618-e608a8c0-13c4-50029-9b5dc-116f1d68-9b5dc;epid=0003D60210DC00
Call-ID: 4b97e08-e608a8c0-13c4-50029-1a8cac-7412a27b-1a8cac
CSeq: 32294 OPTIONS
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
Supported: timer,100rel
Server: IP Office 9.0.5.0 build 972
To: <sip:795@192.168.8.15>
Content-Type: application/sdp
Content-Length: 232

v=0
o=UserA 171425006 3854441077 IN IP4 0.0.0.0
s=Session SDP
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 [highlight #FCE94F]RTP 55010[/highlight]

I'm clear but my fighter is down.

 

[phonesaz]

Typo - disabling 8021q. So the phone works, but there is a noticeable delay before dial tone is extended when the phone goes off-hook; maybe 1.5-2 seconds? And when a call is terminated, there is a delay of 4-5 seconds before a second call can be made. I am not sure what is causing this.