[Asterisk-video] h263 blackholed.

Miles Scruggs asterisk at wideideas.com
Wed May 21 16:36:03 CDT 2008


Do you know of a way to get eyebeam/xlite to announce codecs right out  
the door?


On May 21, 2008, at 2:26 PM, Sergio Garcia Murillo wrote:

> Hi,
>
> Try adding videosupport=yes also to sip peers and if you're using  
> Xlite set it to send the invite
> with video at first and not re-invite latter with video offer (If i  
> recall correctly from the pastebin).
>
> Best regards
> Sergio
>
> Miles Scruggs escribió:
>>
>> Damn it looks like my pastebin has expired but we were only sending
>> one codec, the only thing that would make it fail was to send the
>> codec to asterisk after the call is setup.  If we setup the call with
>> h263 listed then it would work out of the door, but if the call was
>> setup without the codec, then a second invite came down the pipe to
>> asterisk with the new codec (h263) asterisk would pretend to
>> renegotiate but never passed h263.
>>
>> On May 21, 2008, at 2:00 PM, Borja Sixto wrote:
>>
>>
>>> Hi, the standard asterisk do not support cleanlly the sdp  
>>> negociation.
>>> Try setting only one video codec.
>>>
>>> Regards,
>>>
>>>
>>> Tech from i6net
>>>
>>> ----- Message d'origine -----
>>> De: Gunnar Hellstrom <gunnar.hellstrom at omnitor.se>
>>> Env: mercredi 21 mai 2008 19:47
>>> À: 'Development discussion of video media support in Asterisk' <asterisk-video at lists.digium.com
>>>
>>> Objet: Re: [Asterisk-video] h263 blackholed.
>>>
>>> You must use a late VIDCAPS to get any usable video support from
>>> Asterisk.
>>> VIDCAPS adds important ability to handle SDP properly in the call
>>> setup.
>>>
>>> Gunnar
>>>
>>> -----Original Message-----
>>> From: asterisk-video-bounces at lists.digium.com
>>> [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Miles
>>> Scruggs
>>> Sent: Tuesday, May 20, 2008 11:37 PM
>>> To: asterisk-video at lists.digium.com
>>> Subject: [Asterisk-video] h263 blackholed.
>>>
>>> I'm trying to get asterisk to proxy h263 for a video call, but not
>>> having
>>> any luck.  I have posted a full call trace here:
>>>
>>> http://pastebin.com/d330aecb5
>>>
>>> While watching a full packet dump on the asterisk node, I can see  
>>> the
>>> h263 coming in from the clients, but it never leaves (asterisk never
>>> originates 1 h263 packet).  You can see on line 123 of the pastebin
>>> that the
>>> calls get setup without video but then h263 is renegotiated on line
>>> 456.
>>> The second end point sets up h263  on line 313.  I'm
>>> pretty lost as to how I can get asterisk to play nice.  This is
>>> Asterisk 1.4.18.1
>>>
>>> I just was able to test, and if the call sets up (the initial invite
>>> contains) with h263, and isn't done with a re-invite at a later time
>>> then it
>>> works fine.
>>>
>>> my sip conf looks like the following:
>>>
>>> [general]
>>> type=friend
>>> host=dynamic
>>> context=incoming
>>> srvlookup=yes
>>> canreinvite=no
>>> videosupport=yes
>>> qualify=yes
>>> nat=yes
>>> host=dynamic            ; This device registers with us
>>> disallow=all ;better for custom-tunning codec selection allow=ulaw
>>> allow=alaw
>>> allow=h263 ; H.263 is our video codec
>>> allow=h263p ; H.263p is the enhanced video codec
>>> dtmfmode=rfc2833 ; inband is not supported in compressed codecs like
>>> gsm, so
>>> we better set it to rfc2833
>>>
>>>
>>> [miles_8920_lap]
>>> type=friend
>>> host=dynamic
>>> secret=*****
>>> context=internal_devices        ; the internal_devices context
>>> controls what we can do
>>> mailbox=8920 at sixsquares
>>> callerid= Miles Scruggs <8920>
>>> canreinvite=no
>>>
>>> [jon_8926_desk]
>>>
>>> type=friend
>>> host=dynamic
>>> secret=*****
>>> context=internal_devices
>>> mailbox=8926 at sixsquares
>>> callerid=Jonathan Larsen <8926>
>>> canreinvite=no
>>>
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>>
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>
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