[Asterisk-video] Lifesize HD X10 endpoints & Video Resolution and framerate configuration question

[Klaus Darilion]

IIRC there is a new media architecture currently implemented in trunk. 
Maybe they handle media attributes correctly.

There is also an old 1.4 branch by Olle which supports media attributes.

regards
klaus

On 13.04.2011 10:16, Sergio Garcia Murillo wrote:
> Try as much as you like, but it is not going to work :)
>
> Asterisk does not include any attribute of the incoming INVITE to the
> outgoing INVITE, so you will loose the profile-id and the h264 will be
> only established in CIF. Enable the sip logs and verify it.
>
> Best regards
> Sergio
>
>
> El 13/04/2011 1:19, Joel Wiramu Pauling escribió:
>> Hrm, setting either directmedia=yes and directrtpsetup=yes in
>> sip.conf does not seem to fix the issue.
>>
>> I wonder if this is a network issue, everything is on routable
>> address's endpoint wise, and the gateway in between routes between my
>> RFC1918 address network (which the asterisk server sits on), i've done
>> this sort of setup in prod before tho and it works well.
>>
>> On 13 April 2011 10:39, Joel Wiramu Pauling<joel at aenertia.net> wrote:
>>> Cheers will give that a go, thanks for your input Gunnar.
>>>
>>>
>>> wrt @amit: Codec is supported, it's the SDP/ATV combination ( I assume
>>> that's the resolution ) that it is saying is unsupported - h264 ( the
>>> codec for video ) is working fine, I think you are seeing the Siren
>>> (audio) mismatches thats fine it falls back to ulaw.
>>>
>>>
>>> Kind regards
>>>
>>> -JoelW
>>>
>>>
>>> On 13 April 2011 08:35, Gunnar Schaller<linux at nowin.de> wrote:
>>>> Hello,
>>>> Try a Dial without "tr" parameters and with "directmedia=yes" in
>>>> sip.conf.
>>>> http://www.voip-info.org/wiki/view/Asterisk+SIP+media+path
>>>>
>>>> Regards,
>>>> Gunnar
>>>>
>>>>
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>
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