[Top][All Lists]

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

Re: [Linphone-developers] Trouble with RFC 2833 / 4733 DTMF

From: Russell Treleaven
Subject: Re: [Linphone-developers] Trouble with RFC 2833 / 4733 DTMF
Date: Fri, 4 May 2018 14:54:56 -0400


I looked at my response from last night and I can't even tell you what I meant to say :).

Yeah looks to me like like linphone doesn't like dtmf with an iana number of 98.
The extremely typical case is 101.
can you disable speex 16k but leave speex 8k and see if anything changes?

Hope I make more sense today.

On Fri, May 4, 2018 at 2:40 PM, Dominic <address@hidden> wrote:
Hi Russell,

I posted the SDP for both a successful and a failing call. Were
you able to see them both?

To answer your question, yes, voice works fine in both cases.
Linphone and asterisk agree on PCMU/8000 (ulaw). It doesn't look
like linphone offers that codec as an option in the initial
INVITE even though it is enabled in the settings, but it's listed
(a=rtpmap:0 PCMU/8000) in the 200 OK. For some reason, it looks
like the formatting of my last message got messed up in the
email. It's a lot easier to read on the FreePBX forum post if you
want to check it out there:

You said that Asterisk wasn't indicating support for rfc-2883
dtmf or speex. I can confirm that speex is disabled in my
Asterisk codec settings, but it should support rfc-2883 dtmf.
Isn't that what the "a=rtpmap:98 telephone-event/8000" part of
the 200 OK response means? I apologize, I'm still quite new to

> On Thu, 3 May 2018 20:08:32 -0400 Russell Treleaven <address@hidden> wrote:
> Asterisk is not indicating support for rfc-2883 dtmf 16000khz or speex.
> What does linphone think was agreed upon?
> Does voice work in this specific case?

Thank you,

Linphone-developers mailing list


Russell Treleaven

reply via email to

[Prev in Thread] Current Thread [Next in Thread]