How To Solve Common Problems with VOIP and Fire Alarm Systems Dialing Out – Part 2Print This Post
This is part two of solving problems with VOIP and fire alarm system panels. The first part can be viewed here http://www.bohack.com/2012/04/how-to-solve-common-problems-with-voip-and-fire-alarm-systems-dialing-out-part-1/ I covered several common problems like POTS lines, on-hook voltage and VOIP quality settings.
In this second part I will cover in depth transmit volume or gain, receive volume of gain and out of band handling of DTMFs for Contact ID. These are all common problems with VOIP and fire alarm systems. The system ATA or Analog Telephone Adapter I will focus on is an Adtran with AOS, but these principles can be applied to all ATA devices. I still recommend a backup line of a POTS circuit since nothing replaces a simple POTS line, when you are dealing with life safety equipment.
TX Gain and RX Gain
TX gain or transmit volume and RX gain or receive volume can be an issue with fire, alarm, call boxes and any other type of analog panel. These are simple the volumes at which the information is transmitted or received from your panel. The defaults for an Adtran are a TX gain of -6.0 and an RX gain of -3.0 and this should be fine for most voice applications. However when it comes to data, panels can be picky as to what they expect.
One way to identify a problem is to listen to the panel as it is dialing out and the conversation between the transmitter (you) and the receiver (monitoring station). You can achieve this with a simple line’s man butt set, across the terminals of the phone line on the panel. A better way to identify problems with TX and RX gain is to packet capture the conversation and decode it. This will be covered in depth in part 3 of this series.
To raise or lower the TX and RX gain on an Adtran Total Access 924 you can use these commands:
Adtran #conf t Adtran (config)#interface fxs 0/1 Adtran (config-fxs 0/1)#rx-gain 0 Adtran (config-fxs 0/1)#tx-gain -3.0
Be careful not to turn the gain up too high, since you will over gain the transmitter or receiver. This will work opposite of what you are trying to accomplish, so do it in small increments. The defaults are an rx-gain of -3.0 and a tx-gain of -6.0, so it’s safe to increase it in +3.0 increments.
DTMF and VOIP
DTMF tones are Dual Tone Multi Frequency tones that dial numbers. To understand what the function of an ATA is we must understand the process of a phone. When you go off hook you receive a dial tone, where you use the 12 digit key pad of 0 thru 9 and special dial plan characters like star and pound. This series of tones are picked up by the ATA and turned into a phone number to dial.
With SIP calls there are two distinct communication channels. The SIP channel is a symmetrical port 5060 UDP conversation with your SIP trunk. The RTP channel is what transmits your voice and that is usually carried over port 10000-20000 UDP with a typical Asterisk server. The SIP control channel sets up the RTP channel, and the RTP channel uses this to its advantage to transmit data like DTMF digits.
Many fire alarm panels communicate with DTMF tones. So once the call is setup any digits that are sent to the ATA will be intercepted. The DTMF digits sent are then removed from the RTP stream and are pushed to the SIP channel. This operation is called out of band DTMF relay and is the default of many ATAs. Take notice that at time of dial the RTP channel has not yet been established, this is only after the RTP in established. Sometimes the interception and omission of DTMFs is not clean and tones bleed over on the RTP stream, which is the root of our problem.
The default behavior of an Adtran Total Access 924 is to send DTMFs out of band in the SIP channel. This means that the SIP provider recreate the DTMFs on their end to the upstream provider, from the digits provided over the SIP channel. This is process is often known as DTMF Relay and an Adtran by default is configured with the command “rtp dtmf-relay nte 101”. You can change this behavior with the following command:
Adtran #conf t Adtran (config)#voice user 4125551212 Adtran (config-4125551212)# rtp dtmf-relay inband
This command will tell the TA-924 not to intercept the DTMFs and to send them over the RTP channel. However the DTMFs sent for initial call setup will still travel over the SIP channel, since an RTP is not yet established at this point.
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The next article in this series can be found here http://www.bohack.com/2012/04/how-to-solve-common-problems-with-voip-and-fire-alarm-systems-dialing-out-part-3/