[Freeswitch-users] IVR codec issue (I think)

Thu May 22 08:46:52 MSD 2014

Excellent write-up but I have one small correction. RFC2833 is another
payload type in the same RTP stream as the audio. It follows RTP, not SIP.

On Wednesday, May 21, 2014, Guillermo Ruiz Camauer <grcamauer at gmail.com>
wrote:

> AVT seems to be another way of referring to RFC2833.  This is a way of
> sending the DTMFs out of band.  That is, through a separate data stream
> than the voice.  Just like SIP carried the signaling and RTP carries the
> voice.
> RFC 2833 has several advantages over INBand DTMFs.  INBand DTMFs must be
> first extracted from the voice stream and then interpreted, which consumes
> much more CPU than just reading a number.  The DTMF decoders will also get
> confused some times with people with high pitched voices and generate
> spurious DTMFs.  Also, if the RTP packets follow a different route that the
> SIP packets, your FS server will not be able to listen for DTMFs once the
> call has been established (because the RTP packets are not passing through
> FS).  RFC 2833 signaling goes together with SIP, and therefore will always
> be available to FS, so it will be able to monitor key presses and react
> accordingly.
> So if bandwidth is not a problem, use G711u, set Provess AVT to yes, and
> DTMF Tx method to AVT (only) on the Ciscos, and RFC2833 on the FS side sand
> you'll be fine.
>
> Guillermo
>
>
>
> Sent from my iPhone
>
> On 21/05/2014, at 21:33, Sean Devoy <sdevoy at bizfocused.com<javascript:_e(%7B%7D,'cvml','sdevoy at bizfocused.com');>>
> wrote:
>
>  Hi Guillermo,
>
>
>
> Thank you for your response.  I wish understood more, but let me tell what
> I know and ask you another question.
>
>
>
> My (voice) ISP only supports “ULaw, ALaw, GSM and G729a”
>
> My phones are all Cisco504G’s.  They support “G711u, G711a, G729a, G722,
> G726-16 (-24, -24 and -32).  Then there are 2 other items that must be
> relevant: “DTMF Process AVT: Yes or No” and “DTMF Tx method: Inband, AVT,
> Info, Auto, Inband + Info, AVT + Info”
>
>
>
> I think Tx Method defaulted to AVT+Info as I got warnings about receive
> DTMF in INFO on FS.  I am set to AVT and DTMF is working fine.
>
>
>
> Here is what I think (I would know, but it may not be correct) about
> codecs:
>
> G711u = ULaw, G711a=ALaw, The phones don’t seem to support GSM (?) and my
> VISP doesn’t support 722 or any 726. So that leaves me with G711a, G711u,
> G729.  I have abundant bandwidth everywhere involved.  I know G729 is
> fairly high quality compression with regard to bandwidth vs audio quality.
> I also know I have to get a license for each G729 channel on FS for about
> $10 each.
>
>
>
> I do not understand the difference in G711u and G711a.
>
>
>
> What is my best choice for DTMF Tx Method?  And codec?
>
>
>
> Thank you for your time.
>
> Sean
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>[
> mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>]
> *On Behalf Of *Guillermo Ruiz Camauer
> *Sent:* Wednesday, May 21, 2014 7:53 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] IVR codec issue (I think)
>
>
>
> If possible, try to use RFC-2833 for the DTMF and you shouldn't have any
> codec problems.  711u is only required if you are doing INBAND DTMF.
>
>
>
> Guillermo
>
>
>
> On Wed, May 21, 2014 at 6:35 PM, Ian McMaster <ian.mcmaster at gmail.com<javascript:_e(%7B%7D,'cvml','ian.mcmaster at gmail.com');>>
> wrote:
>
>  Try:
>
>
>
> export data="absolute_codec_string=PCMU"
>
>
>
> before the answer in the dialplan.
>
>
>
> After answer, the codec is already established.
>
>
>
> Ian.
>
>
>
>
>
>
>
> On Wed, May 21, 2014 at 5:22 PM, Sean Devoy <sdevoy at bizfocused.com<javascript:_e(%7B%7D,'cvml','sdevoy at bizfocused.com');>>
> wrote:
>
>   <image001.gif>
>
> HI Everybody!
>
>
>
> I had a recent problem where a transferred call’s audio was highly
> degraded.  The cause was identified as mismatched codecs – thanks again.
>
>
>
> Today, I have a VERY VERY similar problem.  The difference is that the
> first step is an IVR, not the receptionist.  How do I set the CODEC for an
> incoming call to 711u when it is routed to an IVR.
>
>
>
> Thanks,
>
> Sean
>
>
>
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> --
> Guillermo Ruiz Camauer
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
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