[Standards-JIG] FW: Jingle - P2P and PBX calls
twaffle at gmail.com
Thu Jan 5 21:51:16 UTC 2006
On 1/5/06, Simon Guindon <simon.guindon at tomahawk.ca> wrote:
> 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?
Personally, I'd use info. This makes the RTP implementation simpler. I
can think of no logical reason NOT to do it over XMPP.
> 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?
This sort of call control is always handled by using a (re)INVITE. After
thinking about it, this is a problem with the correct specification. There
really IS no direct mapping for an INVITE or reINVITE. INVITE can also be
used to renegotiate, and really, this isn't addressed at all.
This sort of functionality would be, I suspect part of 'replace', which,
I'd like to point out, isn't defined. The above conversation can perhaps
help solidify the usage of the replace action within jingle.
Hypothetically, with replace, you'd send a replace to place a user on hold,
or send one when you wanted to call to be on hold.
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