Problems with sending RFC2833 DTMF

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Thanks for sharing :)

Indeed sounds a good candidate. I changed it in my config_site.h and I'll
keep you in touch about users feedback.

As it was introduced by revision 3345 : (
http://trac.pjsip.org/repos/changeset/3345/)
I'm wondering if there is not something else to change :
PJMEDIA_RTP_PT_START has been change to (PJMEDIA_RTP_PT_DYNAMIC-1), instead
of 102.

If we change telephone event type to 101, we may go out of the limits of
dynamic payload type.
Why dynamic payload type not computed when building SDP? (I'm not sure
that's possible but as far as I understood reading SDP docs it should be).
Cause most of the time users/device will not send SDP offers with a long
choice but pjsip will probably support more and more codecs. And having a
static list may break the 127 limit really quickly.
Or at least in codecs maybe it could be interesting to add some conditions
to defs. (For example PJMEDIA_HAS_INTEL_IPP or PASSTHROUGH for AMR and so
on).


2011/1/13 mayamatakeshi <mayamatakeshi at gmail.com>

>
> On Tue, Jan 11, 2011 at 3:19 AM, Alexei Kuznetsov <eofster at gmail.com>wrote:
>
>> Hi,
>>
>> Has anything been changed in sending RFC2833 DTMF between pjsip 1.6
>> and 1.8.5? Our users started complaining about DTMF not working with
>> Google Voice. The previous version of the app that was linked against
>> pjsip 1.6 works fine.
>>
>>
> I was using an even older version of pjsip (1.5.5).
> When I updated to 1.8 I got some RFC2833 issues.
> I verified that the PT for telephone-events have changed at
> pjmedia/include/pjmedia/config.h:
>
> # 1.5.5:
> #  define PJMEDIA_RTP_PT_TELEPHONE_EVENTS          101
> #  define PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR      "101"
>
> # 1.8:
> #   define PJMEDIA_RTP_PT_TELEPHONE_EVENTS          96
> #   define PJMEDIA_RTP_PT_TELEPHONE_EVENTS_STR      "96"
>
> So have it modified back to use 101 and I don't have issues anymore.
>
> br,
> takeshi
>
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