[asterisk-dev] frame size

[Fernando Lombardo]

I have tried to apply the patch attached to bug id 5162, but the behavior
appears to be the same (I'm using sip channels).  When doing an RTP debug I
still see g.729 packets with len=20 (not 40) and g.723 packets with len=24
(not 48). I tried posting my findings to the bug log, but I'm unable to get
a login name so I'm posting here. Is there any configuration that needs to
happen besides putting "packetization = xx" in the peer definitions?
Also, any reason why there is no definition for g.723 in rtp_codec_table
RTP_CODEC_TABLE[]?
Thanks,

-fernando

On 1/21/06, Dan Austin <Dan_Austin at phoenix.com> wrote:
>
> Bug id 5162 makes it possible, at least for G729.  It should apply to
> 1.2.X
> It hasn't been reviewed or accepted into trunk, but works very well for
> us.
>
> Dan
>
>  ------------------------------
> *From:* asterisk-dev-bounces at lists.digium.com [mailto:
> asterisk-dev-bounces at lists.digium.com] *On Behalf Of *Fernando Lombardo
> *Sent:* Saturday, January 21, 2006 7:05 AM
> *To:* asterisk-dev at lists.digium.com
> *Subject:* [asterisk-dev] frame size
>
> Hello, I've been experimenting with Asterisk and transcoding. I have a
> question. Is there a way to make it so that the RTP is sent in 40 byte
> frames (for g.729) or 48 byte frames (for g.723).I noticed when doing a
> rtp debug in Asterisk that it is actually sending 20 byte frames for both
> g729/g723 while the other end sends 40 bytes/48bytes respectively. This
> would cause asymmetric bandwidth consumption. Thanks,
>
> -fernando
>
>
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--
-fernando
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