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List Info
Thread: help for videocall
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| help for videocall |
  Italy |
2008-02-22 02:40:58 |
Hi list,
first of all: sorry for my bad english...
I'm new to Asterisk world... I'm using Asterisk-1.4.13 on
Debian distribution.
I've tested with success loopback, mp4play, mp4save and IVR
calling Asterisk with a 3G mobile phone.
Now I'm trying a 3G<->sip videocall (using a NEC-phone
and X-lite running on Windows) but I think I'm confused
about how to do this... I've read the mailing list archive,
the pages on VoIP-info and the instructions on Fontventa but
I don't understand how to write my dialplan, especially the
order of the following exten...
- h324m_gw()
- h324m_gw_answer()
- h324m_call()
- Dial()
I found contradictory configuration examples reading the
archive...
And do I need the following exten?
exten => dial,1,Set(CHANNEL(transfercapability)=VIDEO)
exten =>
dial,n,NoOp(transfer=${CHANNEL(transfercapability)})
exten => dial,n,Set(CHANNEL(userinformationlayer1)=38)
exten =>
dial,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
...or maybe I mistakes configuring my X-lite...
Thanks very much for any help!
Barbara B.
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| Re: help for videocall |

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2008-02-22 08:35:16 |
barbara_b02 libero.it wrote:
> Hi list,
> first of all: sorry for my bad english...
> I'm new to Asterisk world... I'm using Asterisk-1.4.13
on Debian distribution.
> I've tested with success loopback, mp4play, mp4save and
IVR calling Asterisk with a 3G mobile phone.
> Now I'm trying a 3G<->sip videocall (using a
NEC-phone and X-lite running on Windows) but I think I'm
confused about how to do this... I've read the mailing list
archive, the pages on VoIP-info and the instructions on
Fontventa but I don't understand how to write my dialplan,
especially the order of the following exten...
>
> - h324m_gw()
> - h324m_gw_answer()
> - h324m_call()
> - Dial()
>
> I found contradictory configuration examples reading
the archive...
>
> And do I need the following exten?
>
> exten =>
dial,1,Set(CHANNEL(transfercapability)=VIDEO)
> exten =>
dial,n,NoOp(transfer=${CHANNEL(transfercapability)})
> exten =>
dial,n,Set(CHANNEL(userinformationlayer1)=38)
> exten =>
dial,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
This is needed for SIP-->3G calls (to make 3G video
signalling in ISDN
SETUP message), but not needed for 3G-->SIP calls.
To bridge 3G-->SIP calls:
1. Make sure to have AMR codec installed (and configured
codecs.conf ->
see readme from AMR codec)
2. in sip.conf allow videocalls and active h263p for the SIP
peer
3. MAke proper bandwidth settings in XLITE/eyebeam:
deactivate the "zero touch bandwidth detection"
and then in the
advanced section set the bandwidth to "DSL,
Cable".
4. In eyebeam/xlite active the H263+ codec
5. For testing video calls it might also be useful to
deactivate the
echo canceler in xlite.
6. Don't bother with outgoing calls unless incoming calls
are working fine.
regards
klaus
>
> ...or maybe I mistakes configuring my X-lite...
>
> Thanks very much for any help!
> Barbara B.
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
a>
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
a>
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| Re: help for videocall |
  Spain |
2008-02-22 12:04:20 |
On Fri, 2008-02-22 at 15:35 +0100, Klaus Darilion wrote:
>
> barbara_b02 libero.it wrote:
> > Hi list,
> > first of all: sorry for my bad english...
> > I'm new to Asterisk world... I'm using
Asterisk-1.4.13 on Debian distribution.
> > I've tested with success loopback, mp4play,
mp4save and IVR calling Asterisk with a 3G mobile phone.
> > Now I'm trying a 3G<->sip videocall (using a
NEC-phone and X-lite running on Windows) but I think I'm
confused about how to do this... I've read the mailing list
archive, the pages on VoIP-info and the instructions on
Fontventa but I don't understand how to write my dialplan,
especially the order of the following exten...
> >
> > - h324m_gw()
> > - h324m_gw_answer()
> > - h324m_call()
> > - Dial()
> >
> > I found contradictory configuration examples
reading the archive...
> >
> > And do I need the following exten?
> >
> > exten =>
dial,1,Set(CHANNEL(transfercapability)=VIDEO)
> > exten =>
dial,n,NoOp(transfer=${CHANNEL(transfercapability)})
> > exten =>
dial,n,Set(CHANNEL(userinformationlayer1)=38)
> > exten =>
dial,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
>
> This is needed for SIP-->3G calls (to make 3G video
signalling in ISDN
> SETUP message), but not needed for 3G-->SIP calls.
>
> To bridge 3G-->SIP calls:
> 1. Make sure to have AMR codec installed (and
configured codecs.conf ->
> see readme from AMR codec)
>
> 2. in sip.conf allow videocalls and active h263p for
the SIP peer
>
> 3. MAke proper bandwidth settings in XLITE/eyebeam:
> deactivate the "zero touch bandwidth
detection" and then in the
> advanced section set the bandwidth to "DSL,
Cable".
>
> 4. In eyebeam/xlite active the H263+ codec
>
> 5. For testing video calls it might also be useful to
deactivate the
> echo canceler in xlite.
>
> 6. Don't bother with outgoing calls unless incoming
calls are working fine.
>
> regards
> klaus
>
>
> >
> > ...or maybe I mistakes configuring my X-lite...
> >
> > Thanks very much for any help!
> > Barbara B.
> >
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
> >
> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-video
a>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
a>
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
a>
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| Re: help for videocall |
  Spain |
2008-02-22 12:04:18 |
On Fri, 2008-02-22 at 15:35 +0100, Klaus Darilion wrote:
>
> barbara_b02 libero.it wrote:
> > Hi list,
> > first of all: sorry for my bad english...
> > I'm new to Asterisk world... I'm using
Asterisk-1.4.13 on Debian distribution.
> > I've tested with success loopback, mp4play,
mp4save and IVR calling Asterisk with a 3G mobile phone.
> > Now I'm trying a 3G<->sip videocall (using a
NEC-phone and X-lite running on Windows) but I think I'm
confused about how to do this... I've read the mailing list
archive, the pages on VoIP-info and the instructions on
Fontventa but I don't understand how to write my dialplan,
especially the order of the following exten...
> >
> > - h324m_gw()
> > - h324m_gw_answer()
> > - h324m_call()
> > - Dial()
> >
> > I found contradictory configuration examples
reading the archive...
> >
> > And do I need the following exten?
> >
> > exten =>
dial,1,Set(CHANNEL(transfercapability)=VIDEO)
> > exten =>
dial,n,NoOp(transfer=${CHANNEL(transfercapability)})
> > exten =>
dial,n,Set(CHANNEL(userinformationlayer1)=38)
> > exten =>
dial,n,NoOp(ul1=${CHANNEL(userinformationlayer1)})
>
> This is needed for SIP-->3G calls (to make 3G video
signalling in ISDN
> SETUP message), but not needed for 3G-->SIP calls.
>
> To bridge 3G-->SIP calls:
> 1. Make sure to have AMR codec installed (and
configured codecs.conf ->
> see readme from AMR codec)
>
> 2. in sip.conf allow videocalls and active h263p for
the SIP peer
>
> 3. MAke proper bandwidth settings in XLITE/eyebeam:
> deactivate the "zero touch bandwidth
detection" and then in the
> advanced section set the bandwidth to "DSL,
Cable".
>
Xlite will probably send too much video bitrate so probably
you'll see
video lagging and delaying from audio. In this case use
app_transcode to
reencode the video to the correct bitrate.
Best regards
Sergio
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
a>
|
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| Re: help for videocall |
  Spain |
2008-02-22 12:10:10 |
On Fri, 2008-02-22 at 09:40 +0100, barbara_b02 libero.it
wrote:
> Hi list,
> first of all: sorry for my bad english...
> I'm new to Asterisk world... I'm using Asterisk-1.4.13
on Debian distribution.
> I've tested with success loopback, mp4play, mp4save and
IVR calling Asterisk with a 3G mobile phone.
> Now I'm trying a 3G<->sip videocall (using a
NEC-phone and X-lite running on Windows) but I think I'm
confused about how to do this... I've read the mailing list
archive, the pages on VoIP-info and the instructions on
Fontventa but I don't understand how to write my dialplan,
especially the order of the following exten...
>
> - h324m_gw()
> - h324m_gw_answer()
> - h324m_call()
> - Dial()
>
h324m_gw is the function to receive incoming videocalls from
the ISDN.
It creates a pseudo channel back into the dialplan with the
extension
and context specified in the parameter.
This local channel has to be answered by h324m_gw_answer,
which is just
like answer but signalling a VIDEOUPDATE to the h324_gw
application so
no video is lost while setting the call. You can also use
Dial instead
to call the SIP softphone.
h324m_call is for outgoing videocalls.
Best regards
Sergio
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-video mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-video
a>
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