Audio (RTP or UDP) traffic routed incorrectly

[Randall Garner]

Our ISP provides us with voice and data services - each service has its own ethernet connection - data is supposed to stay on the data connection and voice is supposed to stay on the voice connection. The ISP describes the voice service as a Peer-to-Peer application of a SIP trunk.

 

Our PBX has two NICs - one connected to the voice connection (for the voice service to communicate with the PBX) and the other connected to our LAN (for the VoIP phones to communicate with the PBX). Also note our PBX is running Windows Server 2008.

 

During a recent failure of our internet firewall, not only did we lose data (as expected), but was also lost the audio in our calls (not expected). During the internet outage, we still received rings, but no audio from the callers could be heard.

 

It appears the PBX is sending and receiving the signaling from the voice connection, but is routing all of the actual voice traffic over the data connection - which is not how the service is designed.

 

As best I can tell, Windows is able to communicate/route traffic correctly - it appears the PBX is the culprit.

 

This system was setup about a year ago by another provider - he had much trouble getting the solution to work and has now told us he has no wish to help us correct the setup of the system he installed.

 

Can you please help me locate the correct settings so that all voice (signaling and audio packets) are correctly routed over the voice connection?

[Vodia PBX]

We tried the same thing (with Linux) in our own office. It is a difficult topic.

 

The way to differentiate RTP from other traffic is the DiffSrv bits. You can set them in the PBX up (take a look at the pbx.xml file). The default values should already be okay.

 

For inbound traffic, whoever is sending the traffic needs to know where to send the packets, that must be done by the IP address. The consequence is that you need to assign a specific IP address for the RTP which is different than the IP address for the SIP, HTTP and other protocol ports.

 

There is a setting "Bind to specific IP address" where you can specify to what IP address the PBX should bind the RTP ports (for IPv4 and IPv6, respectively). This should solve the problem that all RTP traffic will be send from and to a specific IP address, which you can connect to the special line for RTP.

 

We tried some time ago with Linux, but were not able to tell the OS to stick to the IP address when sending traffic out. Maybe thats easier in Windows.

[Randall Garner]

I located the "Bind to IPv4 Address" setting under Ports -> RTP. There, I found the LAN IP of the PBX entered.

 

Is this setting possibly causing the RTP traffic to route out of the internet connection on the LAN, and not allowing RTP to route over the SIP connection (the other NIC in the PBX)?

[Vodia PBX]

Probably. I would definitevely make a backup of the PBX directory, so that in case anything gets screwed up, you can move back. Then find a time when there is no traffic and then you can try to change the setting. With changes to the network configuration you have to be a lot more careful as it is not so easy to undo them later.

 

Also, I would try to paint a diagram on how it should look like, so that the changes you do on the routing make sense.

[Randall Garner]

I made the backup of the Snom One folder, then removed the LAN IP of the PBX from the Ports -> RTP -> Bind to IPv4 setting. Saved the setting, then restarted the Snom One PBX Serivce.

 

Used Wireshark to observe the SIP/RTP traffic afterwards and now appears correct.

 

Thank you for the help!