[asterisk-users] Early Media Issue

[Mark Farmer]

It's a good shout but sadly hasn't helped. Thanks anyway!

The issue seems to be that our provider expects to be able to send inband
early media.
There is an OpenSIPS box between the provider & Asterisk which essentially
just routes SIP traffic so the behaviour at our end is still controlled by
Asterisk which makes the call.

Using dtmfmode=auto it seems to be possible to switch to inband if RFC2833
is not advertised in SDP but the provider just honours what we set in the
call setup, which, since we only use RFC2833 is always advertised in SDP.

ATM I think it's a provider issue, according to another environment they
should never send us inband but it seems to not be working correctly in the
case.

Regards
Mark.


On Mon, 17 Jun 2019 at 10:11, Floimair Florian <f.floimair at commend.com>
wrote:

> Just a guess, but I suspect that this might be related with strictrtp
> setting in rtp.conf, which learns the correct source in doing so drops a
> few packets.
>
> I would try to disable strictrtp for testing purposes and if this works
> add some delay before playing back the media.
>
>
>
>
>
>
>
> With best regards
>
>
>
> *Florian Floimair *Innovation - Software-Development
>
>
> *COMMEND INTERNATIONAL GMBH *A-5020 Salzburg, Saalachstraße 51
> http://www.commend.com
>
>
>
> *Security and Communication by Commend *FN 178618z | LG Salzburg
>
>
>
> *Von: *asterisk-users <asterisk-users-bounces at lists.digium.com> im
> Auftrag von Mark Farmer <farmorg at gmail.com>
> *Antworten an: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Datum: *Freitag, 14. Juni 2019 um 15:15
> *An: *Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users at lists.digium.com>
> *Betreff: *[asterisk-users] Early Media Issue
>
>
>
> Hi all
>
>
>
> I've got an issue where when I call a number that just plays early media
> back to me.
>
> Instead of hearing the full sequence of tones I hear a short ringing then
> part of the sequence. What seems odd is that I can see
> the telephone-event/8000 being passed up the chain but when it gets to
> Asterisk, it is never sent back to the phone. Instead I just see the usual
> RTP flows.
>
>
>
> I've been trying to fix this for hours, does anyone have any ideas how to
> get this working correctly?
>
>
>
> Asterisk version is 13.25.0
>
>
>
> The settings I think are relevant (I'm using chan_sip):
>
>
>
> (sip.conf)
>
> ignoresdpversion=yes
>
> internal_timing=yes
>
> progressinband=never
>
> silencesuppression=no
>
> prematuremedia=no
>
>
>
> (Per peer)
>
> progressinband=yes
>
> directrtpsetup=no
>
> dtmfmode=rfc2833
>
> directmedia=no
>
> silencesuppression=no
>
> prematuremedia=no
>
>
>
>
>
> TIA
>
> Mark.
>
>
>
> --
>
> Mark Farmer
> farmorg at gmail.com
> --
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-- 
Mark Farmer
farmorg at gmail.com
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