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Re: [vt-dev] Re: SECN-1 with Softphone Support


Keith Williamson Apr 7, 2012 6:06 PM
Posted in group: Village Telco Development Community
Hi Glen, Elektra (and all),

So this turns out to just work (at least as far as my testing went this afternoon). I just changed canreinvite from "no" to "yes" and nat from "yes" to "no" in the two entries in softphone.sip.conf used for my EVO smartphone (running linphone) and my Toshiba tablet (running 3CXphone).

The calls between them worked just great and I verified that the RTP media stream was NOT going through the MASTER asterisk server (in Asterisk console, "rtp set debug on"). Prior to the change, console shows constant stream of RTP packets, after change, RTP packets stop after a few at the beginning of the call.

I also tested calls between MP and softphone and got same result! So it appears that Asterisk on the MP accepts the redirected media stream by default.

Note that I didn't need to set "directmedia=yes" as that's the default setting for that. Will still need to ensure we haven't broken the ability for the softphones to make calls through an external VSP (if configured). If so, there are a few globals we can add to the sip config:

directmediadeny=0.0.0.0/0
directmediaaccept=10.130.1.0/24  /* Change for local network address

This only allows the media to be redirected to p2p if both SIP endpoints are on the local network.

I'll leave it up to others further testing and consensus as to whether we want to sneak this change into SECN1.1 RC3. It would only affect the softphone.sip.conf fixed configuration file.

Cheers,

Keith



On Sat, Apr 7, 2012 at 5:39 PM, Glen Steedman <gste...@gmail.com> wrote:
Hi Keith,

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



On 8 April 2012 06:08, Keith Williamson <hkwill...@gmail.com> wrote:
> This is definitely an area we want to investigate for improved
> softphone support for SECN2.0. This should work fine for Softphone to
> Softphone calls however looks more troublesome for Softphone to MP
> calls. I don't believe the MP's ATA entity (UA) was designed to
> operate independently of it's local asterisk instance but perhaps
> Elektra can comment. In other words, can the MP ATA access the L2 mesh
> network like any other SIP client? That would be pretty cool but I
> don't think it works that way.
>
> Still the Softphone <--> Softphone calls should be able to use
> reinvite to talk peer-to-peer. That has great potential for improving
> call quality, and more evenly distributing mesh network load which
> should result in improved scalability.
>
> Thanks for opening the topic!
>
> Cheers,
>
> Keith
>
> On Apr 5, 5:46 am, Sjur Eivind Usken <s...@usken.no> wrote:
>> > Are there any plans with the softphones to allow the Media to be sent to
>> > the softphones directly ?
>>
>> you can just set that in Asterisk.
>>
>> check the following settings (in sip.conf file, either as global variables
>> or on each sip profile)
>>
>> reinvite = yes
>> nat =  no
>>
>> and the
>> directmedia = yes
>>
>> my asterisk knowledge is a little outdated, but these should do it...
>
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