DTMF end of event

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Hi Benny,

I had provided a patch for DTMF end of event issue. I have attached that email communication with this mail. I was trying version pjsip 1.10 and I do not see this patch. Is this issue resolved? Has the code been modified to take care of this issue? So far, I have been using my patch and it is working great.

Once again thanks for the great software.

Madhuri

[pjsip] New features: audio latency, NAT, PRACK/UPDATE, AKA,	etc.Madhuri Patwardhan madhuri_patwardhan at yahoo.com 
Tue Nov 27 05:31:20 EST 2007 
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________________________________
 
Hi Benny,

I have attached the file stream.c.patch. This was generated with diff aganist pjsip version 0.8.0. I have also attached the modified stream.c.
Hope it is useful.

Madhuri


----- Original Message ----
From: Benny Prijono <bennylp@xxxxxxxxx>
To: pjsip embedded/DSP SIP discussion <pjsip at lists.pjsip.org>
Sent: Friday, November 23, 2007 8:29:03 PM
Subject: Re: New features: audio latency, NAT, PRACK/UPDATE, AKA, etc.


Madhuri Patwardhan wrote:
>Hi Benny,
>
>Sorry for the late reply.
>
>My patch sends 3 end of event packets for each dtmf digit
 irrespective of the next digit in the queue. Perhaps, I am missing something but
thats what I see sjphone and other voip clients doing. 3 end of event
packets are sent irrespective of the next dtmf digit in the queue. The
remote end client to which we are connecting using pjsip recommends using
3 end of event packets.

I think you're right. Although the text in RFC 2833 somehow implies 
that retransmissions should only be done when there's no new event 
in the queue, I found the clarification in RFC 4733, which obsoletes 
RFC 2833:

2.5.1.4.  Retransmission of Final Packet

The final packet for each event and for each segment SHOULD be
sent a total of three times at the interval used by the source
for updates.

Could you send me the diff again?

cheers,
-benny

>Madhuri
>
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