I have started a wish list of items that either need testing or developed. I am going to make an attempt to at the very least provide further debug and/or testing data, but I would like to see the following issues get solved and commited to the Asterisk svn tree.

SIP Spiral is a situation when Asterisk and OpenSER are used together. This situation occurs when Asterisk, as a PSTN media gateway, receives a call and sends it to OpenSER; in turn OpenSER sends the call back out to Asterisk for PSTN termination. In other words, a very typical call forwarding situation. I have worked around the SIP Spiral issue by having dedicated ingress and egress Asterisk media gateways, but now there is code that is supposed to correct Asterisk’s behavior, but it needs further testing.

Dan York is going to be speaking about VoIP Security and mentioned Asterisk SRTP. There has been a patch out there for a while now, which we tested at Interop 2007. We will attempt to test SRTP as well, provided we can acquire the necessary, compatible SRTP enabled VoIP equipment. We have access to Grandstream phones and I know somebody there will have Counterpath. With a little luck we should be able to document what works and what does not work with the current level of Asterisk SRTP skills.

The trouble we had at Interop 2007 was that not every VoIP device vendor supports the same key exchange algorithms. Thus, we were forced to use SDES, which is already not being favored by the IETF. Also, in order to be truly secure, Asterisk needs to support TLS, which means SIP TCP support - in other words there are very significant changes required to the SIP channel driver.

Perhaps with a little luck, the appropriately skilled people will be attending Astricon this year and just maybe we could motivate the SIP TCP and TLS support at least a little farther along to at least create a patch for further testing after the show.

Another big feature request is T.38. Unfortunately the current featureset of Asterisk (not to mention patents) are not quite there yet for Asterisk to become a T.38 endpoint (meaning converting the udptl frames into modem tones.) However, Asterisk can now pass-through the data allowing for those VoIP devices that supports T.38 to talk via Asterisk to a VoIP Provider that supports T.38.

We are planning on demonstrating the T.38 pass-thru feature by using a Grandstream ATA using a regular fax machine and sending the calls out to NuFone using the hotel’s network. (again, with more luck)

Lastly, I would like to find time to finish adding RealTime to Music On Hold, even though I am not a very big fan of Asterisk realtime. By adding in the realtime hooks into MOH, one will have a bit more flexibility, especially on those systems that are already utilizing realtime.

Please feel free to comment or otherwise contact me with your ideas on what we should test or work on in the CodeZone at Astricon 2007.

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