Where to start: reduce network packet rate

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Hi,

There is a SDP "ptime" attribute that you can add to your SDP, but AFAIK
this is just merely an indication to peer about our preference ptime. The
peer may or may not adhere to this. Can't you buffer the packets and
process them at reduced rate?

Best regards,
 Benny


On Wed, Mar 5, 2014 at 8:54 PM, Jeroen Hoppenbrouwers <
jHoppenbrouwers at avionica.com> wrote:

>  Hi all,
>
>
>
> For an embedded project using uClinux on a virtual CPU at 80 MHz, we
> deploy PJLIB/PJSIP as the audio transport. It sits in between our codec
> hardware (16-bit 8000 Hz no frills) and a standards-compliant SIP/RTP
> network link over Ethernet. The SIP link may go directly to the peer (the
> same software on another box) or be routed through Asterisk for
> conferencing. Three of these bi-directional audio links can be active at
> the same time on the box.
>
>  It all works, but we have issues with the many Ethernet/UDP/RTP packets
> that flow around. Our minimal hardware has trouble coping with the sirq and
> CPU load. A very crude hack suggested that halving the RTP packet rate by
> doubling the content significantly reduced the problem. However I have
> trouble cleanly setting up the SDP negotiation to ask the peer for
> double-content/half-rate. I can fix myself, but not the other side, so to
> say.
>
>
>
> Current packet length (ptime?) is 20ms, 320 bytes (160 samples of 16
> bits). I'd like to get to 640 or 1280 bytes instead (MTU is 1500 bytes).
> Latency is less of a problem in our application.
>
>
>
> Q: where do I need to start looking in the PJLIB stack for this setting,
> and what am I actually looking for exactly? Is this done at the codec
> layer, at the pjmedia layer, at the SIP toplevel layer? Etc.
>
>
>
> Best regards,
>
>
>
>
>
> Jeroen Hoppenbrouwers
>
>
>
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>
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>
>
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