Olle has posted a note explaining his with for codec negotiation fixup:

Friends,

On my wishlist for 1.8 long-term-support release the #1 item is a new media negotiation platform for Asterisk.

If we had integrated John Martin's videocaps in time for release of 1.4 we now would have enjoyed four releases of an Asterisk with much better video support, instead of the broken support we have today. Integration of that code was denied because of plans of something greater and bigger, which we have discussed for years at many Astridevcons. Because of us denying this and other code, outside-of-tree code has been around that fixes a lot of issues and the original Asterisk still lacks proper video support.

There's also a few codec negotation patches that has been around and maintained for years, one of the major ones being Maxim Sobolev's patch.

There are a few main issues here:

• Codecs should not be handled as a bit, being turned on and off. Both audio and video codecs have attributes that we need to take care of properly
• When answering, the properties of the answer needs to be relayed to the calling channel, not just a control frame that says "somebody answered".
• We should be prepared for multiple media streams, including multiple media streams of the same format

The new solution should be extensible, it should be easy to add both new codecs and properties.

The solution has to be configurable. The way you want Asterisk to set up a bridge between two channels varies much. Some people prefer Asterisk doing transcoding some people want Asterisk to stay out of as much trouble as possible and just set up whatever is most simple. Other users just want to standardize all call legs to one type of phone, but have a different policy for other connections. There's no one-solution-fits-all.

We did write up a few documents on this a year ago at Astridevcon in Phoenix. I would really like to see some work on this. My personal feeling is that this is very important for the continued success of Asterisk in the marketplace. The amount of long-lived patches in this area that has been maintained for years shows that we have to do something (and that we sadly have ignored customer demand). The arrival of new codecs and new solutions, like video conferences with multiple audio and video channels, with text channels and possibly other types of channels (lika a binary channel for MSRP and digital ISDN) - all this tells us that we have to get there.

I haven't seen Digium invest in this project during the years we have discussed this. The Digium team has fixed the codec list, it is no longer a limited bitmap. Building a new media negotiation framework not a simple project and it's not something any customer alone would fund, it would propably allocate too much resources from the team. (my personal guesses) I don't think the community as a whole can expect or demand that Digium funds every needed change themselves, especially not in these times of financial worry. Everyone in the eco system will have to contribute to make Asterisk a better product - and I'm not only talking money here.

Maybe we have to apply for funding somewhere else. It requires much more knowledge of Asterisk and experience than what I think we can get through Google summer-of-code, and it's more urgent. Let's discuss this and see how we can make this happen. Read the docs, start an open discussion and hopefully we can get not all of this, but at least the core of it, inside the next long term release of Asterisk, 1.8. It will require a lot of job from all of us in the dev team, as well as the community. Testing, feeding input, building a stabile architecture. I am sure we can get it done.

The Astridevcon documentation is at:
http://astridevcon.pbworks.com/Media+Negotiation

It includes a lot of thoughts going down to code structures.

My presentation that was a base for the discussion is at
http://svnview.digium.com/svn/asterisk/team/group/astridevcon2008/NewMediaArchitecture.pdf

The floor is open. What do you think?

/Olle

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The Asterisk Development Team has announced security releases for Asterisk as the following versions:

• 1.6.0.22
• 1.6.1.14
• 1.6.2.2

These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The releases of Asterisk 1.6.0.22, 1.6.1.14, and 1.6.2.2 include the fix described in security advisory AST-2010-001.

The issue is that an attacker attempting to negotiate T.38 over SIP can remotely crash Asterisk by modifying the FaxMaxDatagram field of the SDP to contain either a negative or exceptionally large value. The same crash will occur when the FaxMaxDatagram field is omitted from the SDP, as well.

For more information about the details of this vulnerability, please read the security advisory AST-2009-009, which was released at the same time as this announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.22
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.1.14
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.2

Security advisory AST-2010-001 is available at:

http://downloads.asterisk.org/pub/security/AST-2010-001.pdf

Thank you for your continued support of Asterisk!

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Hi all, I was going through the stats for the Daily Asterisk News and noted that the star rating system I was using was taking 180KB to download.

I've now replaced it with an onChange select box - less than a k.

Hopefully it's pretty easy to use, drop me a note or post a comment if it doesn't make sense :D

If you've got any comments as to how we can make the Daily Asterisk News a more useful resource, just let me know.

Comments (2 posted)

Written by Nicki (February 4, 2010, 9:18 am)

Nice to see some people still care about how much bandwidth something uses :-)

Bandwidth
Written by Matt Riddell - http://www.venturevoip.com (February 4, 2010, 1:14 pm)

:) It does make a difference - snappiness of the site, usability on a mobile platform etc :)

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Julian Lyndon-Smith has posted details of a patch he has written to temporarily mute MixMonitor recordings:

I have uploaded a patch for 1.4 and trunk that allows you to mute either or both parts of a mixmonitor recording. I would appreciate it if someone apart from me could test it and let me know how you get on.

Thanks!

Julian

https://issues.asterisk.org/view.php?id=16740

for PCI-DSS compliance we are not allowed to record a credit card number is a MixMonitor file. However, we must record all conversations
....

I have added a new feature to audiohooks so that you can mute either read / write or both types of frames - this allows for MixMonitor to mute either side of the conversation without affecting the conversation itself.

MixMonitor now has two manager commands

1) manager show command MuteMixMonitor

Action: MuteMixMonitor
Synopsis: Mute a channel in MixMonitor
Privilege: <none>
Description: Mutes a Mixmonitor Channel.
Variables:
Channel: Channel to mute.
Direction: Which part to mute. read|write|both (from channel|to channel|both channels).

2) manager show command unMuteMixMonitor

Action: unMuteMixMonitor
Synopsis: unMute a channel in MixMonitor
Privilege: <none>
Description: unMutes a Mixmonitor Channel.
Variables:
Channel: Channel to unmute.
Direction: Which part to unmute. read|write|both (from channel|to channel|both channels).

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Olle has posted an update on IPV6 in Asterisk and a link to a blog post of his:

Friends,

Before the Christmas holidays, I did send this letter and did not get a lot of response, but some. Since then, I've been able to get interest from a few parties that are willing to fund parts of this work, including Digium, the main sponsor of Asterisk. I will also apply for additional funding from a foundation here in Sweden and hope to get some more responses so that we can fund this project together. If anyone out there has interest or feedback regarding IPv6, Asterisk and VoIP, I'll be happy to get in contact.

I've documented some of my thoughts on how to proceed, based on the work already done by Marc Blanchet (and of course work together with him) on my blog, http://www.voip-forum.com/asterisk/2010-01/voip-users-care-ipv6/

My hope is that we can get this done and integrated in Asterisk 1.8, but that requires some immediate attention from the community, as well as help with testing and feedback when we start rolling. Marcs code is already out there, so you can start testing NOW in your IPv6-enabled network. http://www.asteriskv6.org/

IPv6 is a boring topic, and if you do it right, no one will thank you for it. It just needs to be done. My work with IPv6 started the summer of 1995 and since then people have been shouting "We need to migrate now!". We've done that so long so that no one listens any more and now it's getting really critical. The IP numbering authorities, like ARIN and RIPE, have already outlined how they will have to change procedures for IPv4 assignments every six months from now, making it harder and harder to get addresses. For VoIP - sip trunks, calling each other across the Internet, it's critical to have public IP addresses unless you want to stay with your lovely Telco on the other end of the copper cables.

Personally, I'm not sure how to design software for this migration properly. In order to educate myself and collegues that develop and build SIP solutions, I'm going to organize an event this spring which combines testing and training. I do hope that the Asterisk community will join me and support the developer team in our efforts to make Asterisk - the leading Open Source PBX - running perfectly well on both IPv4 and IPv6 networks. It needs to be done, we will get it done. And no one will thank us for it, since everyone just expects Asterisk to work as we have done for the last 10 years...

With IPv6 greetings!

/Olle

Comments (1 posted)

Training You Say?
Written by Roberto Guerra (February 4, 2010, 1:14 pm)

IPv6 VoIP training? I am interested. Is it going to be in Alabama?

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