asterisk users - Oct 2004 - Amazing, great protocol IAX

Hi;

I've just had a couple of discoveries in the learning process that are
really making me impressed with this software.  

1) IAX transfer - I'm running Asterisk boxes A, B, and C.  B is in the
middle and has a dialplan that points to extensions on C.  When a client
on A in the proper context on B tries dialling a client of C, B is smart
enough to release itself somehow from playing telephone in the middle,
letting A and C network directly with each other.  This totally rocks. 
Who cares if it takes an extra 2 - 3 seconds to complete dialling.  This
is called Native Transfer Mode, no?  I don't see why anyone would want
to disable it unless they were worried about missing CDR's - or hadn't
figured out how to connect to a remote DB from A, C, or both.  

2) Absolute control over codecs and translations.  This took me a while
to understand how to force things to get translated.  I'd be curious to
know a couple of pointers on this subject - 

A) Which is better sound quality in this situation?
On a local net (100 megabit) my SIP phone talks to Asterisk running on
my firewall.  I only have 128K upload (sometimes) on my fricken
Third-World DSL connection so I want to use GSM or ILBC going out over
IAX2.  Should I let the DSP in the phone talk to asterisk in ULAW or
ALAW, having Asterisk translate to GSM or should I force the phone to do
it in the lower bandwidth codec?  I honestly can't tell a difference in
quality or latency.  I'm not real impressed with the Grandstream
Handytone 286 because it doesn't have GSM - so I'm stuck trying to
compare straight through ILBC to translated GSM.  Any fine tuning codec
advice would be appreciated.

B) I forgot what I was thinking for B, sorry about that.. lost my...

C) Going back to the possibility of losing CDRs, I can see another
possible time when a person would want to disable Native Transfer Mode -
when the IAX client is an IAXY box. (no way they write CDRs, let alone
cdr_postgres)  I'm still itching to try one of those things out -
anybody have good or bad experiences with them?

Thanks, much obliged,

Thomas Hutton

On Sun, 03 Oct 2004 22:04:46 -0300, Thomas Hutton <pres@nicheware.com>
wrote:
> 
> 1) IAX transfer - I'm running Asterisk boxes A, B, and C.  B is in the
> middle and has a dialplan that points to extensions on C.  When a client
> on A in the proper context on B tries dialling a client of C, B is smart
> enough to release itself somehow from playing telephone in the middle,
> letting A and C network directly with each other.  This totally rocks.
> Who cares if it takes an extra 2 - 3 seconds to complete dialling.
It doesn't take any longer to dial. The transfer happens about 8-10
seconds into an already established call.
> is called Native Transfer Mode, no?  I don't see why anyone would want
> to disable it unless they were worried about missing CDR's - or
hadn't
> figured out how to connect to a remote DB from A, C, or both.
In some cases where both B and C are behind a NAT, transfer will have
to be disabled because the hosts cannot see each other directly.
> Should I let the DSP in the phone talk to asterisk in ULAW or
> ALAW, having Asterisk translate to GSM or should I force the phone to do
> it in the lower bandwidth codec?
Depends on how limited your CPU capacity on the Asterisk server is. If
you run into CPU bottlenecks because of transcoding, then you are
better off to let the phone use the low bandwidth codec so Asterisk
can simply pass it through without doing transcoding.
> quality or latency.  I'm not real impressed with the Grandstream
> Handytone 286 because it doesn't have GSM - so I'm stuck trying to
> compare straight through ILBC to translated GSM.  Any fine tuning codec
> advice would be appreciated.
ILBC is definitely superior in sound quality to GSM.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

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Thomas Hutton wrote:

  > A) Which is better sound quality in this
situation?
> On a local net (100 megabit) my SIP phone talks to Asterisk running on
> my firewall.  I only have 128K upload (sometimes) on my fricken
> Third-World DSL connection so I want to use GSM or ILBC going out over
> IAX2.  Should I let the DSP in the phone talk to asterisk in ULAW or
> ALAW, having Asterisk translate to GSM or should I force the phone to do
> it in the lower bandwidth codec?  I honestly can't tell a difference in
> quality or latency.  I'm not real impressed with the Grandstream
> Handytone 286 because it doesn't have GSM - so I'm stuck trying to
> compare straight through ILBC to translated GSM.  Any fine tuning codec
> advice would be appreciated.
> 
Why do you need GSM at all, just keep everything iLBC. Asterisk won't 
have to transcode and it will sound much better

-Adam

On Sun, 3 Oct 2004, Thomas Hutton wrote:
> 1) IAX transfer - I'm running Asterisk boxes A, B, and C.  B is in the
> middle and has a dialplan that points to extensions on C.  When a client
> on A in the proper context on B tries dialling a client of C, B is smart
> enough to release itself somehow from playing telephone in the middle,
> letting A and C network directly with each other.  This totally rocks. 
> Who cares if it takes an extra 2 - 3 seconds to complete dialling.  This
> is called Native Transfer Mode, no?  I don't see why anyone would want
> to disable it unless they were worried about missing CDR's - or
hadn't
> figured out how to connect to a remote DB from A, C, or both.  
It rocks even more than you think.  Before and whilst the transfer is
taking place, audio is still passed via the B system.
> A) Which is better sound quality in this situation?
> On a local net (100 megabit) my SIP phone talks to Asterisk running on
> my firewall.  I only have 128K upload (sometimes) on my fricken
> Third-World DSL connection so I want to use GSM or ILBC going out over
> IAX2.  Should I let the DSP in the phone talk to asterisk in ULAW or
> ALAW, having Asterisk translate to GSM or should I force the phone to do
> it in the lower bandwidth codec?  I honestly can't tell a difference in
> quality or latency.  I'm not real impressed with the Grandstream
> Handytone 286 because it doesn't have GSM - so I'm stuck trying to
> compare straight through ILBC to translated GSM.  Any fine tuning codec
> advice would be appreciated.
If all your calls go through to the DSL line, you'd probably do best using
iLBC on your phone.  Asterisk doesn't have to transcode, and your phone
has probably got iLBC's lost packet concealment, which isn't implemente
on
Asterisk right now.

G711 does sound better - so if you will have calls to other LAN places or 
use Asterisk IVR, Voicemail etc a lot, then maybe its worth using G711.

Regards,
Steve

PS: Where in the 3rd world?