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

Sergio Garcia Murillo sergio.garcia at fontventa.com
Wed Apr 13 03:56:40 CDT 2011


Hi klaus, good to see you here;)

That are good news, anyone know what are the new functionalities 
provided by this new media architecture?

Best regards
Sergio

El 13/04/2011 10:30, Klaus Darilion escribió:
> 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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