Are you trying to originate the call from
freeswitch (requiring freeswitch to actually send the multicast rtp stream) or
are you originating from another device that will send the multicast rtp and
create the proper sdp, but will send through freeswitch to handle the sending
of invites to all the parties?
If freeswitch is going to originate the
stream, what is the source of the audio they will be sending to all the calls? I
assume that you get no return audio from any of the end devices?
Mike
From:
freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
On Behalf Of Sluschny, Thomas
Sent: Wednesday, February 21, 2007
11:20 AM
To:
Subject: Re: [Freeswitch-users]
Special SIP scenario
Thank you Mike for the answer.
But:
i would not only call multiple
parties/devices/receivers, but also announce all these receivers with one
sender at the same time!
To do this the audio data is multiplied in
the network, thats why we use (ip-) multicast: see below c=IN IP4 239.194.0.14
And this is also a problem because the
"c=" attribute contains normally the own IP and not a multicast one,
so: is it possible to set some (all?) SDP
attributs directly?
(i saw e.g. remote_media_ip in http://wiki.freeswitch.org/wiki/FreeSwitch_Channel_Variables (under
Unknown Functionality)
Thomas
Von:
freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
Im Auftrag von Michael Jerris
Gesendet: Mittwoch, 21. Februar
2007 15:59
An:
Betreff: Re: [Freeswitch-users]
Special SIP scenario
The originate syntax allows for multiple calls
out with “and” (,) and “or” (|) logic. This
should be documented on the wiki for originate syntax. This should also
allow you to play a file to the answering party and do key confirmation to
initiate a bridge with the original calling party. As far as custom
codecs, either for passthrough only, or with transcoding, you would need to
make a codec module, at least setup to do the passthrough, the amr and g729
codecs in tree are a good example of this. Typically, you should not ever
use codec numbers in the static range (<97 I believe) unless they are
officially assigned, that being said, freeswitch will not keep you from writing
a module that will work for this. If freeswitch is truly not in the media
path at all, you can use no media mode, which would pass through the sdp raw to
the other side, in which case you would not even need the passthrough codec, as
log of course as the endpoints knew how to handle the multicast.
Mike
From:
freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
On Behalf Of Sluschny, Thomas
Sent: Wednesday, February 21, 2007
9:02 AM
To:
Subject: Re: [Freeswitch-users]
Special SIP scenario
Hello,
unfortunately no one answered til now. So
i ask a bit different:
1. Is it possible to create a call to a
SIP device where i can define the MessageBody or SDP field on my own (as
mentioned in first mail)?
I want to INVITE one sender (plays an
announcement) and many receivers. These INVITEs should be independent, so if
one receiver is out of order all others can be announced anyway.
2. Is it possible to invite a SIP device
independend from others?
I tested:
pacall device1; pacall device2;
uuid_bridge sip_uuid1 sip_uuid2 -> this is a little bit sick, i know ;)
originate sofia/domain/device1 device2
-> but this works only with 2 devices AND the INVITEs go out successively
Has this "media" command
anything to do with it?
Hope this clears more my problem.
Thanks,
Thomas
Von:
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Sluschny, Thomas
Gesendet: Donnerstag, 15. Februar
2007 17:42
An:
Betreff: [Freeswitch-users]
Special SIP scenario
Hi all,
I want to do a special SIP using and I dont know how to configure it
with: originate/bridge/transfer no_media and so on
May
be it is not possible with freeswitch?
Here comes scenario:
-
one sender device that sends multicast RTP stream to network
-
one or more receiver devices that only receive RTP data from network
The
codec is a proprietary one (41), freeswitch can ignore all RTP data.
Sender has to get attributs:
u=file:/C:/dap/123.wav
c=IN
IP4 239.194.0.14
t=0
0
m=audio
7280 RTP/AVP 41
a=rtpmap:41
PCMA/16000
a=recvonly
b=AS:128
Receiver has to get attributs:
c=IN
IP4 239.193.201.69
t=0
0
a=specialExtension:2
a=type:broadcast
m=audio
7280 RTP/AVP 41
a=rtpmap:41
PCMA/16000
a=sendonly
Is it possible to use freeswitch API to do this?
Many thanks in advance,
Thomas
Sluschny