Are there any plans with the softphones to allow the Media to be sent to the softphones directly ?
not sure I understand the problem right.
The IP-to-IP calling feature in the MP dialplan works independently from any server requirement. Same goes for softphones, if they have a independent IP-to-IP dialling feature. All I have to do with a softphone like twinkle is to set up a profile for IP-to-IP calls and dial sip:4000@MP-IP. Same goes for making calls in the opposite direction. The Asterisk extension in the MP initiates a call to sip:4000@Softphone-IP if you dial XXX or XXX*XXX*XXX*XXX, so the caller id configuration in the softphone should be set to 4000.
Calls go directly - so the mesh takes the best routing path and there is no server required. Besides: All versions of BATMAN can route asymmetrically if that is the best option.
As far as I understand, RTP traffic flows directly between the phones. A SIP server is only required to initiate the call. Take a look at this picture:
http://de.wikipedia.org/w/index.php?title=Datei:SIP_signaling.svg&filetimestamp=20111003154001
But I might be missing.
Cheers,
Elektra
one solution might be to run a stripped down version of freeswitch which
can scale down to being a "softphone" to make the MP act as a gateway.
this will require a voip server somewhere (preferably inside the mesh)
to connect the gateway to.
Michel
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I didn't test calling from the MP(ata) set to client, but im assuming
it just routing calls to ext 30x to 30x@Master_Ip_Address_252
which no doubt would mean the RTP path is from MP Ata, to Master then
back again to the softphone. In cisco speak the Sip Proxy can operate
in media flow around (endpoints directly with RTP) or flow through
(RTP hairpins in the proxy server) Usually the later is used for any
codec conversion, dtmf conversions , security / qos etc etc
so keep with the same example
MP33 ATA calls Softphone 300 , RTP would hairpin all the way to the
Master MP (04 in this example) Rather then staying on MP33 where the
Softphone and ATA are.
Ill do some more testing next week tweaking the media options from
Sjur's post and see what works (and what stays and doesn't stay after
a reboot)
I guess another option could be, if its going to be a static
deployment of SoftPhones you could then hardcode the IP address to the
Softphone, then modify the DialPlan of the MP's to call the IP address
of 300 directly without going through the Master MP ( assuming the
Softphone client will accept a direct sip invite, some may only accept
it from the sip proxy they are registered too)
Cheers, Glen