One Way Audio Suggestions

If you are on a “bring your own device” (BYOD) plan with your VoIP provider, you will have access to the settings in your analogue terminal adapter (ATA). This can come in handy if the phone of the person you are calling rings but you do not hear a ring tone, or if you experience the common one-way audio problem where the person you have called answers the call and can hear you OK, but you cannot hear them at all.

It is essential to understand that VoIP calls are a two stage process. Firstly there is signaling which is related to call setup (connecting the two endpoints). Then there is the audio portion, which is the actual voice call where the two parties can speak to each other. The second thing to understand is that the internal port number of your ATA is “mapped” to a different port number by your router and the remote device must respond to the mapped port number for things to work smoothly.

The missing ring tone issue is all to do with internal (or private) IP addresses and external (or public) IP addresses being used in the packets being exchanged between the two ATA’s. Any IP addresses from inside your private network are not routable on the public Internet and are therefore useless to any devices outside your network (the Internet side of your router).

Some, but not all VoIP ATA’s are clever enough to learn what their external IP address and their mapped SIP port number are by inspecting the responses they receive from registration packets they send to the VoIP provider server. On Sipura/Linksys adapters, there are two settings named “Handle VIA received” and “Handle Via rport” that must be set to yes for this “learning mode” to be enabled. The result is that when a call is made to a remote adapter, the external IP address and mapped port number of your ATA are placed inside the VIA line of the INVITE packet rather than the internal address and SIP Port. This ensures the call setup packets are returned to the correct routable IP address and mapped SIP port of the calling ATA and a ring tone should be heard.

Although the above technique overcomes the missing ring tone issue, it does not help with one way audio problems. Without the help of your VoIP provider server, you will require the use of a STUN server. STUN is not only a good alternative way for your ATA to “learn” it’s external IP address and mapped SIP port number, but it also allows it to learn it’s mapped RTP Port – which is critical for achieving two-way audio.

A VoIP ATA will normally have settings for two RTP port numbers. These define the range of ports that the ATA may dynamically choose to use for RTP – the audio portion of a VoIP call. One way audio is caused where the “un-mapped” RTP port is inserted into the SDP portion of an INVITE packet. The called ATA will send RTP packets to that port number and they will be rejected by the caller’s router.

The use of a STUN server overcomes this issue and helps your ATA insert the correct mapped RTP port into the SDP portion of an INVITE packet. The result is that the call is setup successfully and the called ATA returns the RTP stream to the correct port on the caller’s router.

By: IP Alarms

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Steve Nutt is the owner of Sovereign Monitoring Systems, Yeoman Monitoring Services and partner with Camwatch IP in the UK.