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

[Sergio Garcia Murillo]

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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