[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


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 at pjsip.org>
To: pjsip embedded/DSP SIP discussion <pjsip at lists.pjsip.org>
Sent: Friday, November 23, 2007 8:29:03 PM
Subject: Re: [pjsip] 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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