I’ve created a short guide on how SIP manages audio streams and the sorts of things that go wrong when those streams traverse NATs. The full guide can be read at SIP and Audio Guide.

To complement the guide I’ve whipped together a diagnostics tool.

SIPSorcery RTP Diagnostics Tool

In an attempt to help people diagnose RTP audio issues I have created a new tool that provides some simple diagnostic messages about receiving and transmitting RTP packets from a SIP device. The purpose of the tool is twofold:

  1. On a SIP call indicate the expected socket the RTP packets were expected from and the actual socket they came from,
  2. On a SIP call indicate whether it was possible to transmit RTP packets to the same socket the SIP caller was sending from.

To use the tool take the following steps:

  1. Open http://diags.sipsorcery.com in a browser and click the Go button. Note that the web page uses web sockets which are only supported in the latest web browsers, I’ve tested it in Chrome 16, Firefox 9.0.1, Internet Explorer 9,
  2. A message will be displayed that contains a SIP address to call. Type that into your softphone or set up a SIPSorcery dialplan rule to call it,
  3. If the tool receives a call on the SIP address it will display information about how it received and sent RTP packets.

The tool is very rudimentary at this point but if it proves useful I will be likely to expend more effort to polish and enhance it. If you do have any feedback or feature requests please do send me an email at aaron@sipsorcery.com.

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