Posted: Sun May 27, 2007 8:52 pm Post subject: [Asterisk-video] Newcomer question about Asterisk and videoc
Hello,
A while ago, there were some ongoing work (and a dedicated branch I
believe) to support video capabilities and their associated negociation
in Asterisk.
Does anyone has a status on this? Is it dead ? What are the plan for 1.4
branch and 1.6 ?
Emmanuel
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Posted: Mon May 28, 2007 6:45 am Post subject: [Asterisk-video] Newcomer question about Asterisk and videoc
27 maj 2007 kl. 22.52 skrev Emmanuel BUU:
Quote:
Hello,
A while ago, there were some ongoing work (and a dedicated branch I
believe) to support video capabilities and their associated
negociation in Asterisk.
Does anyone has a status on this? Is it dead ? What are the plan
for 1.4 branch and 1.6 ?
Hello!
1.4 is released an will not change in regards to features.
During the development meeting in Atlanta we spent a lot of time
discussing a new solution for call setup netotiation - it involved
transport (IPv4/IPv6/tcp/udp/tls) as well as codec and media
handling. The videocaps branch is a good proposal to handle the media
part, but there is a lot of more to it. It will also require changes
to the IAX2 protocol.
Hopefully we can use the videocaps branch as a platform and get
something done in time for 1.6.
As soon as we have the notes available from that session, it will be
published and I'll inform the list.
Best regards,
/Olle
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Posted: Mon May 28, 2007 12:07 pm Post subject: [Asterisk-video] Newcomer question about Asterisk and videoc
Olle E Johansson wrote:
Quote:
27 maj 2007 kl. 22.52 skrev Emmanuel BUU:
> Hello,
>
> A while ago, there were some ongoing work (and a dedicated branch I
> believe) to support video capabilities and their associated
> negociation in Asterisk.
> Does anyone has a status on this? Is it dead ? What are the plan for
> 1.4 branch and 1.6 ?
Hello!
1.4 is released an will not change in regards to features.
How compatible are the videocaps modification with Asterisk 1.4.4 ?
Would it be easy to merge them ? Is there a short description of the
change done on the framework.
Quote:
During the development meeting in Atlanta we spent a lot of time
discussing a new solution for call setup netotiation - it involved
transport (IPv4/IPv6/tcp/udp/tls) as well as codec and media handling.
The videocaps branch is a good proposal to handle the media part, but
there is a lot of more to it. It will also require changes to the IAX2
protocol.
Will this change enables the peer to peer media transport while
retaining central call supervision ?
Quote:
(...)
As soon as we have the notes available from that session, it will be
published and I'll inform the list.
Best regards,
/Olle
_______________________________________________
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Posted: Mon May 28, 2007 1:18 pm Post subject: [Asterisk-video] Newcomer question about Asterisk and videoc
28 maj 2007 kl. 14.06 skrev Emmanuel BUU:
Quote:
Olle E Johansson wrote:
>
> 27 maj 2007 kl. 22.52 skrev Emmanuel BUU:
>
>> Hello,
>>
>> A while ago, there were some ongoing work (and a dedicated branch
>> I believe) to support video capabilities and their associated
>> negociation in Asterisk.
>> Does anyone has a status on this? Is it dead ? What are the plan
>> for 1.4 branch and 1.6 ?
> Hello!
>
> 1.4 is released an will not change in regards to features.
How compatible are the videocaps modification with Asterisk 1.4.4 ?
Would it be easy to merge them ? Is there a short description of
the change done on the framework.
Videocaps is based on svn trunk.
Quote:
> During the development meeting in Atlanta we spent a lot of time
> discussing a new solution for call setup netotiation - it involved
> transport (IPv4/IPv6/tcp/udp/tls) as well as codec and media
> handling. The videocaps branch is a good proposal to handle the
> media part, but there is a lot of more to it. It will also require
> changes to the IAX2 protocol.
Will this change enables the peer to peer media transport while
retaining central call supervision ?
>
We will still be a PBX. We do handle peer2peer setup, but will always
require setting up the call in a way so that
Asterisk can take part in the call. That means that we will not
accept unknown media streams in the call setup.
If you need that for SIP, you need a SIP proxy.
/O
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