Hello asteriskers, I hope someone could help me ... !!
I bought a TC400B, and I am testing doing calls with G729 and G723.
When I used G729 it works fine, but when I try to use G723 the RTP has very
low quality and is not possible to hear to the other person in the phone.
I try to find out the problem but the only weird that I saw in the debug
info is this line (using asterisk 1.2.17):
asterisk*CLI> Apr 15 19:42:44 WARNING[17321] chan_zap.c: Frame too large
Apr 15 19:42:48 WARNING[17321]: chan_zap.c:4931 zt_write: Frame too large
This is my scenario:
- 1 Intel(R) Core(TM)2 CPU (Core 2 Duo with 2.13GHz)
- 1 Gigabit Memory
- kernel 2.6.18-1.2798.fc6 x86_64
- OS: Fedora Core 6
I tried first with asterisk 1.4.1 and its dependencies but the problem is
there, too much noise when someone speeaks (poor voice quality).
In asterisk 1.2.17 and its dependencies (libpri, zaptel and addons) is a
little better but the voice quality is not good anyway. In this scenario
appears the Warning : Frame too large.
The show transcoders and show translations looks fine in asterisk CLI:
asterisk*CLI> show transcoder
0/0 encoders/decoders of 92 channels (G.729a / G.723.1 5.3 kbps) are in use.
asterisk*CLI> show translation
Translation times between formats (in milliseconds)
Source Format (Rows) Destination Format(Columns)
g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - 4 1 1 3 3 2 5 2 - 14
gsm 3 - 2 2 2 2 1 4 3 - 13
ulaw 1 3 - 1 2 2 1 4 1 - 13
alaw 1 3 1 - 2 2 1 4 1 - 13
g726 3 3 2 2 - 2 1 4 3 - 13
adpcm 3 3 2 2 2 - 1 4 3 - 13
slin 2 2 1 1 1 1 - 3 2 - 12
lpc10 4 4 3 3 3 3 2 - 4 - 14
g729 2 4 1 1 3 3 2 5 - - 14
speex - - - - - - - - - - -
ilbc 4 4 3 3 3 3 2 5 4 - -
asterisk*CLI>
After of this I upgrade the kernel to 2.6.20-1.2944.fc6 x86_64, and the
problem remains.
Any help will be so much appreciated.
--
Jovanny Saravia
Solutions Manager
e-solutions Ltda
jovanny.saravia@gmail.com
+57-310-7676163
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Jovanny Saravia wrote:> Hello asteriskers, I hope someone could help me ... !! > > I bought a TC400B, and I am testing doing calls with G729 and G723. > > When I used G729 it works fine, but when I try to use G723 the RTP has > very low quality and is not possible to hear to the other person in > the phone. > > I try to find out the problem but the only weird that I saw in the > debug info is this line (using asterisk 1.2.17): > > asterisk*CLI> Apr 15 19:42:44 WARNING[17321] chan_zap.c: Frame too large > Apr 15 19:42:48 WARNING[17321]: chan_zap.c:4931 zt_write: Frame too large >Looks like you are going to have to modify the frame size for G723 on the phones. Not sure what frame size asterisk is expecting but I would guess 20ms or 30ms.> This is my scenario: > > - 1 Intel(R) Core(TM)2 CPU (Core 2 Duo with 2.13GHz) > - 1 Gigabit Memory > - kernel 2.6.18-1.2798.fc6 x86_64 > - OS: Fedora Core 6 > > I tried first with asterisk 1.4.1 and its dependencies but the problem > is there, too much noise when someone speeaks (poor voice quality). > > In asterisk 1.2.17 and its dependencies (libpri, zaptel and addons) is > a little better but the voice quality is not good anyway. In this > scenario appears the Warning : Frame too large. > > The show transcoders and show translations looks fine in asterisk CLI: > asterisk*CLI> show transcoder > 0/0 encoders/decoders of 92 channels (G.729a / G.723.1 5.3 kbps) are > in use. > > asterisk*CLI> show translation > Translation times between formats (in milliseconds) > Source Format (Rows) Destination Format(Columns) > > g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc > g723 - 4 1 1 3 3 2 5 2 - 14 > gsm 3 - 2 2 2 2 1 4 3 - 13 > ulaw 1 3 - 1 2 2 1 4 1 - 13 > alaw 1 3 1 - 2 2 1 4 1 - 13 > g726 3 3 2 2 - 2 1 4 3 - 13 > adpcm 3 3 2 2 2 - 1 4 3 - 13 > slin 2 2 1 1 1 1 - 3 2 - 12 > lpc10 4 4 3 3 3 3 2 - 4 - 14 > g729 2 4 1 1 3 3 2 5 - - 14 > speex - - - - - - - - - - - > ilbc 4 4 3 3 3 3 2 5 4 - - > asterisk*CLI> > > After of this I upgrade the kernel to 2.6.20-1.2944.fc6 x86_64, and > the problem remains. > > Any help will be so much appreciated. > > -- > Jovanny Saravia > Solutions Manager > e-solutions Ltda > jovanny.saravia@gmail.com <mailto:jovanny.saravia@gmail.com> > +57-310-7676163 > >------------------------------------------------------------------------ > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- Andres Technical Support http://www.telesip.net