Re: [asterisk-users] G.722 between Eyebeam and a Polycom IP650

2008-09-28 Thread Thomas Kenyon
Steve Underwood wrote:

 If I were building a terminal, I'd make mine announce 8000, but accept 
 8000 or 16000 to try to maximise compatibility. It seems people don't do 
 that.
 
Looking at debug output from 1.6 (using a grandstream), it looks like 
that is what it does.

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Re: [asterisk-users] G.722 between Eyebeam and a Polycom IP650

2008-09-28 Thread Michael Graves
On Sun, 28 Sep 2008 12:05:06 +0800, Steve Underwood wrote:

   
There is no error in RFC3551. There is a clear statement that an earlier 
RFC did things wrong, due to a typo, and classifying G.722 as an 8000 
sample/second codec is, for better or worse, the standard. Its messy and 
inconsistent, but its the standard.

The only manufacturers I know of who do the wrong thing (i.e. using 
16000 in the SDP) are Grandstream and Aastra. Both are aware that their 
products are incompatible with the rest of the universe, but seem 
uninterested in fixing them.

If I were building a terminal, I'd make mine announce 8000, but accept 
8000 or 16000 to try to maximise compatibility. It seems people don't do 
that.

Unless you have some special version of eyebeam, I don't think it 
supports G.722. It supports G.722.2, but that is completely different. 
It also supports 16 bit PCM and DVI4 at 16000 samples/second.


No, it's the OEM version which I got from ZipDX. The version that
Counterpath sells direct to end-users does not have the same selection
of codecs as the OEM version. You can see the codec list in the
following image from the audio cod
dialogue.

http://www.mgraves.org/voip/wp-content/uploads/2008/09/eyebeamcodecs.png


Are 16 bit PCM and DVI14 supported by any hard phones?

Michael

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http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
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[asterisk-users] G.722 between Eyebeam and a Polycom IP650

2008-09-27 Thread Michael Graves
Hi All,

So I've been exploring the use of G.722 encoded wideband audio
recently. I have three different SIP devices that allow this: Eyebeam,
IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
together. Calls pass between them in what the Polycom notes as HD
mode and the audio quality is certainly very good.

However, things are not so easy with Eyebeam and the IP650. When a call
is placed between those two the audio stream from Eyebeam to the IP650
is never heard. The audio from the IP650 to Eyebeam is heard, and very
good quality.

David Frankel of ZipDX tells me that there is an error in RFC3551 such
that G.722 RTP clock/timestamps are actually wrong. To quote the RFC
directly.

Even though the actual sampling rate for G.722 audio is 16,000 Hz, the
RTP clock rate for the G722 payload format is 8,000 Hz because that
value was erroneously assigned in RFC 1890 and must remain unchanged
for backward compatibility.  The octet rate or sample-pair rate is
8,000 Hz.

It seems that some manufacturers adhere strictly to the RFC while
others correct for the error. As such there are problems with G.722
interoperability.

Counterpath defends their implementation as being according to the RFC.

This begs the question of what does Asterisk do with G.722? I've yet to
try v1.6 so I open the question to the group.

Michael

--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245




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Re: [asterisk-users] G.722 between Eyebeam and a Polycom IP650

2008-09-27 Thread Steve Underwood
Michael Graves wrote:
 Hi All,

 So I've been exploring the use of G.722 encoded wideband audio
 recently. I have three different SIP devices that allow this: Eyebeam,
 IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine
 together. Calls pass between them in what the Polycom notes as HD
 mode and the audio quality is certainly very good.

 However, things are not so easy with Eyebeam and the IP650. When a call
 is placed between those two the audio stream from Eyebeam to the IP650
 is never heard. The audio from the IP650 to Eyebeam is heard, and very
 good quality.

 David Frankel of ZipDX tells me that there is an error in RFC3551 such
 that G.722 RTP clock/timestamps are actually wrong. To quote the RFC
 directly.

 Even though the actual sampling rate for G.722 audio is 16,000 Hz, the
 RTP clock rate for the G722 payload format is 8,000 Hz because that
 value was erroneously assigned in RFC 1890 and must remain unchanged
 for backward compatibility.  The octet rate or sample-pair rate is
 8,000 Hz.

 It seems that some manufacturers adhere strictly to the RFC while
 others correct for the error. As such there are problems with G.722
 interoperability.

 Counterpath defends their implementation as being according to the RFC.

 This begs the question of what does Asterisk do with G.722? I've yet to
 try v1.6 so I open the question to the group.

 Michael
   
There is no error in RFC3551. There is a clear statement that an earlier 
RFC did things wrong, due to a typo, and classifying G.722 as an 8000 
sample/second codec is, for better or worse, the standard. Its messy and 
inconsistent, but its the standard.

The only manufacturers I know of who do the wrong thing (i.e. using 
16000 in the SDP) are Grandstream and Aastra. Both are aware that their 
products are incompatible with the rest of the universe, but seem 
uninterested in fixing them.

If I were building a terminal, I'd make mine announce 8000, but accept 
8000 or 16000 to try to maximise compatibility. It seems people don't do 
that.

Unless you have some special version of eyebeam, I don't think it 
supports G.722. It supports G.722.2, but that is completely different. 
It also supports 16 bit PCM and DVI4 at 16000 samples/second.

Regards,
Steve


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