New SIP features for Asterisk 1.6

In the past week, I have merged patches into Asterisk trunk that provide new features for chan_sip. These features will be available in Asterisk 1.6.

TCP and TLS support

The Commit

In the past, Asterisk has only had support for UDP as a transport for SIP signaling. Asterisk 1.6 will have support for both TCP and TLS, as well.

This work was done by Brett Bryant and James Golovich, who were both funded by Digium for their work.

SIP Session Timers (RFC 4028)

The Commit

Asterisk 1.6 will also have support for SIP Session Timers, as defined by RFC 4028. These changes prevent stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session.

This work was done by Raj Jain. It was funded by John Todd from TalkPlus, Inc., as well as JR Richardson of Ntegrated Solutions.

Both of these features are major steps forward for SIP support in Asterisk!

Zaptel 1.2.23 and 1.4.8 Released

The Asterisk.org development team has released Zaptel versions 1.2.23 and 1.4.8.

These releases contain a number of bug fixes as well as new features, including:

  • New and greatly improved fxotune utility
  • Full support for new Digium cards, AEX2400, TE121, TE122
  • DTMF generator updates to allow tones to be generated at runtime, as well as support for a DTMF “twist”, on a per-zone basis. The tones for Brazil have been updated to include a 2 dB DTMF twist.

These releases are available for immediate download from downloads.digium.com.

Thank you for your support!

JACK interfaces for Asterisk

I just merged a new module that I have been working on into trunk. The module is app_jack. It provides interfaces between Asterisk and JACK (Jack Audio Connection Kit). A description of JACK from their website:

Have you ever wanted to take the audio output of one piece of software and send it to another? How about taking the output of that same program and send it to two others, then record the result in the first program? If so, JACK may be what you’ve been looking for.

JACK is a low-latency audio server, written for POSIX conformant operating systems such as GNU/Linux and Apple’s OS X. It can connect a number of different applications to an audio device, as well as allowing them to share audio between themselves. Its clients can run in their own processes (ie. as normal applications), or can they can run within the JACK server (ie. as a “plugin”).

I recently learned about PureData, so very quickly I wanted to be able to get phone calls in and out of Pd (PureData). After exploring various options, JACK turned out to be the obvious choice for providing audio transport between Asterisk and Pd. I have been wanting to build some interesting applications in Pd that do audio analysis and manipulation, but I had some infrastructure work to complete first.

So, I wrote app_jack. This module provides two interfaces between Asterisk and JACK. The first is an Asterisk application, JACK(). The second interface is a dialplan function, JACK_HOOK().

JACK() Application

This is the simpler of the two interfaces. When the JACK() application is executed in the Asterisk dialplan, two JACK ports get created. There is an input and output port that acts as the endpoint of a phone call. The audio from the channel goes out of the output port that gets created. Whatever audio that comes in on the input port is what gets sent back to the caller. This would allow for some advanced voice applications that interact with the audio of the call and run as a separate application.

Example:

exten => 1234,1,Answer
exten => 1234,n,JACK()

For more information, see the built in documentation for the application from the Asterisk CLI:

*CLI> core show application JACK

JACK_HOOK() Function

This interface is a little bit more complex, but it is the much more interesting one, in my opinion. The JACK_HOOK function creates an audiohook and attaches it to the channel. In this case, instead of the JACK interface being the endpoint of the phone call, it is simply hooked in to the audio path for a phone call to something else. Audio that comes from a caller gets sent out the JACK output port, and whatever audio that comes back in on the input port gets sent on as the caller’s audio. This allows for cool applications that do analysis and manipulation of the audio in a phone call. One example is that I can now write custom vocoders in Pd.

For more information about the syntax for using JACK_HOOK, see the built in documentation from the Asterisk CLI:

*CLI> core show function JACK_HOOK

Here is a simple example for enabling a JACK_HOOK for a call:

exten => _1NXXNXXXXXX,1,Answer
exten => _1NXXNXXXXXX,n,Set(JACK_HOOK(manipulate)=on)
exten => _1NXXNXXXXXX,n,Dial(IAX2/myprovider/${EXTEN})

This also inspired me to write a new CLI command, “core set chanvar”, which lets you set a channel variable or function from the Asterisk CLI. You can use it to turn on a JACK_HOOK for an existing call from the CLI.

*CLI> core set chanvar SIP/poly1-ab23jadf234 JACK_HOOK(manipulate) on

Another way to enable a JACK_HOOK for an existing call would be by using the manager interface. For example:

Action: SetVar
Channel: SIP/123-f234kjsdfz
Variable: JACK_HOOK(manipulate)
Value: on

Enjoy! Please let me know if you find this interesting or useful. I would also be happy to consider any suggestions for enhancements.