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VOIP communication problem

Posted on 2008-06-25
Medium Priority
Last Modified: 2010-04-12
Can someone offer advice with the following:

We have a VOIP system install currently incomplete, there have been various factors responsible for the delay in the completion of the works, broadband issues, firewalls, support delays etc. We are now at a point where we can make VOIP calls from our loccal obx to a remote pbx but while the remote pbx can recieve these internal calls and hear the caller the caller cannot hear any response from the remote pbx. The remote pbx also cannot call the local pbx. The installer informs me that the LAN or the VPN between these pbxs' is now at fault because the packets are being lost.

Q. I can establish remote desktop sessions between the the local and remote offices and vica versa, so imho there is no problem with data packet transfers between the two sites. Am I right in my assumption that if the data packets are the same and that the VPN or the LAN is proven not to be responsible for the loss of data packets as these successful rdp sessions prove so? If not is there a way other than by simply testing by means of a call that the packets are transferring successfully? Is there a difference between the VOIP packets and the packets being sent and recieved during a rdp session?

Question by:chris1307
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Accepted Solution

pierky earned 500 total points
ID: 21863917
I assume you are using SIP protocol and private IP addresses on the LAN.
IMHO the problem is in the LAN firewall/gateway NAT configuration, not in packet loss between the two networks.
I think call-signaling (SIP) and audio-stream (RTP) stop on your firewall/gateway because your IP-phones or VoIP clients use private IP address... anyway it's hard to debug with so few information.
Try to look for "VoIP and NAT-traversal"!
LVL 19

Assisted Solution

feptias earned 500 total points
ID: 21865634
I agree with pierky. It sounds like a NAT traversal issue - one-way audio is a typical symptom of NAT traversal problems:

Linking two sites using a VPN should, in theory, make it much easier to get VoIP communication working between them because the firewall rules can be quite relaxed and LAN-to-LAN via VPN does not have to use NAT. However, the setup of your inter-site VPN will depend on your IT department's policies and preferences. Furthermore, the path used between end points could be more complex than that between two PBX's if you are using IP phones connected to the PBX. ...and the configuration of the PBX's might be quite tricky if they are making connections both through the VPN link and through a NAT firewall via the Internet.

To answer your question about RDP vs. VoIP, the answer is that there is a big difference: RDP uses one port but VoIP uses several - VoIP establishes an initial connection and then uses this to negotiate the port numbers and IP addresses for the additional connections it needs for the audio stream (RTP). Those connections are initiated from both end points, whereas the RDP connection is only initiated from one. As for packet loss - that's bullsh*t.

Author Comment

ID: 21865891
Thanks for input guys.
Question: If the firewalls on both ends are identical/similar in their set-up at both ends and the firewall manufacturer assures me there are no settings restricting or blocking access through the VPN/IPSEC tunnel, how come one end is able at least to recieve calls and have audio but it's not able to dial or broadcast voice to the other end?

Excuse me being a novice but I am trying hard to resolve this and feel I am stuck between the phone installers and the firewall providers playing tennis with responsibility..

LVL 19

Expert Comment

ID: 21866100
Could it be a routing problem? i.e. any device connected to the network will use a routing table (and default gateway) to send IP packets to another address. Assuming your VPN tunnel has different subnets at each end, could it be that the PBX at one end needs it's routing table to be fixed so it knows how to send packets to the other subnet via the VPN?

Can you use a packet sniffer to see what happens when the non-working end tries to initiate a call? If it uses SIP, you should see a packet sent to port 5060 on the remote PBX.

Author Comment

ID: 21866192

Very impressed with knowledge in link.

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