[Freeswitch-users] Special SIP scenario

Wed Feb 21 08:44:30 PST 2007

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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Sluschny, Thomas
Sent: Wednesday, February 21, 2007 11:20 AM
To: freeswitch-users at lists.freeswitch.org
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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Michael Jerris
Gesendet: Mittwoch, 21. Februar 2007 15:59
An: freeswitch-users at lists.freeswitch.org
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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Sluschny, Thomas
Sent: Wednesday, February 21, 2007 9:02 AM
To: freeswitch-users at lists.freeswitch.org
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 at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von
Sluschny, Thomas
Gesendet: Donnerstag, 15. Februar 2007 17:42
An: freeswitch-users at lists.freeswitch.org
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 <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 

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