[Standards-JIG] FW: Jingle - P2P and PBX calls

Simon Guindon simon.guindon at tomahawk.ca
Thu Jan 5 20:36:58 UTC 2006


Thanks for the feedback Thomas!

 

Yes Steve also mentioned to me on IRC that RTP can send DTMF. Is there
any cases where there is a benefit to sending this via XMPP perhaps in a
Jingle INFO or any other similar manner? I'm not sure which is the
better solution. In the future will some audio methods implement Jingle
that may not have DTMF support but may support putting people on hold,
transferring calls etc?

 

Also about the putting people on hold and how SIP manages it. I still
believe we need this support in XMPP if Jingle is to connect to a PBX.
Reason being, it's the PBX that handles the call management, not the
client. If I understand Asterisk correctly, it is the one putting people
on "music on hold" or allowing a client to do call transfers etc. 

 

This means we must send commands to the Jingle-Asterisk gateway to tell
it what to do on a specific call.

 

Or am I misunderstanding something?

 

Thanks,

Simon

 

-------------------------------------------------------

Simon Guindon

Tomahawk Technologies Inc.

simon.guindon at tomahawk.ca

www.tomahawk.ca

-------------------------------------------------------

________________________________

From: standards-jig-bounces at jabber.org
[mailto:standards-jig-bounces at jabber.org] On Behalf Of Thomas Charron
Sent: Thursday, January 05, 2006 1:03 PM
To: Jabber protocol discussion list
Subject: Re: [Standards-JIG] FW: Jingle - P2P and PBX calls

 

On 1/5/06, Simon Guindon <simon.guindon at tomahawk.ca> wrote:

	Jingle INFO message sounds good to me. This way the gateway can
take
	this INFO message and whether it's a SIP, IAX or any other
gateway it 
	can do its specific DTMF method based on the INFO.

 

  Yes.  There is also another alternative, requiring the clients to be
smarter by sending the DTMF data with an RTP informational message,
which the endpoint handling the call must handle.  SIP can handle them
either way, depending on the capabilities of the client. 


 

	I'm not sure if we want to entirely put the logic on the clients
	shoulders. I think this is a good idea for Jingle clients which
will not 
	be hooking up to PBX's who will be getting multiple Jingle
calls. But in
	the case of someone hooking up to Asterisk, Asterisk has to be
signaled
	to put someone on hold, or transfer etc between lines as far as
I
	understand how Asterisk works and I may be wrong, but I believe
Asterisk
	is doing all the switching.

 

  ...  Handle it the same way SIP puts someone on hold.  (re)INVITE them
with an address of 0.0.0.0 (called a null address, basically).
 

	Could a Jingle INFO message be sent to the gateway to signal the
PBX of
	different call management features? 
	I like the idea of clients being able to switch themselves etc
but
	perhaps if the gateway supports a namespace indicating it has
PBX
	support the client must also signal to the gateway its call
changes?

 

  Since there are many discussions on mapping this with SIP, everything
you'd do with SIP is capable of being done over Jingle.

  A good source of information on 'How SIP does it' can be found at
http://www.egyed.com/faq/sip_faq.html


  Thomas Charron

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