[Asterisk-Users] Re: Polycom sound quality problems

[Wiley Siler]

See my other email....

My setup....

VoIP Provider ---> My T1 ---> Asterisk ---> Sip Clients

The only time I get robo-voice is when the latency to the VoIP provider
is high.
Translating from IAX to SIP should not be a problem but maybe it is in
the build you have?

I run COS on my Polycom segment but with 100 meg switches (9GB
Backplane) and 100 MB network, there is little internal latency.

I would look to the external IAX segment.

W



 

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Mason
Sent: Friday, April 01, 2005 9:22 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Polycom sound quality problems

I don't see any way to tell the Polycom to "ignore" QoS.  It's mainly
routers and switches that pay attention to QoS, the phone would just set
QoS on its outgoing packets.  Anyway, here's what's in the QoS section-
it all seems to be related to sending packets:

QoS
RTP
802.1Q User Priority
	
IP ToS Minimize Delay
	Enabled Disabled
IP ToS Maximize Throughput
	Enabled Disabled
IP ToS Maximize Reliability
	Enabled Disabled
IP ToS Minimize Cost
	Enabled Disabled
IP ToS Precedence
	
Call Control
802.1Q User Priority
	
IP ToS Minimize Delay
	Enabled Disabled
IP ToS Maximize Throughput
	Enabled Disabled
IP ToS Maximize Reliability
	Enabled Disabled
IP ToS Minimize Cost
	Enabled Disabled
IP ToS Precedence
	
Other Protocols
802.1Q User Priority
	
The problem is not that it's choppy or breaks up.  Asterisk is connected
to the phone through two 100mbit switches, so throughput isn't a
problem.  It just sounds very distorted, like a cross between a robot
and Donald Duck.

It really seems to be a problem with the way Asterisk is bridging the
call from IAX to the phone.  It does SIP <-> SIP bridges (not
reinviting) just fine.



Noah Miller wrote:
> Hi Eric -
> 
>>>> I'm having a problem with my Polycom phones and hoping someone else

>>>> has experienced the same thing: Outbound calls are fine, and 
>>>> inbound calls originating from another SIP phone are fine, but 
>>>> inbound calls to the Polycom phone from an IAX channel sound like 
>>>> you're talking to a robot.  The person on the Polycom sounds fine 
>>>> to the person on the IAX channel, however.  Inbound calls to our 
>>>> soft phones sound just fine.
>>>>
>>>> Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) 
>>>> Polycom SoundPoint IP500 SIP Sixtel is the IAX provider.
>>>
>>>
>>> Check to see what codec is being used for the call.
>>> Sean
>>>
>> Default is U-law, but I also switched it to A-law with the exact same
>> results.
> 
> 
> I might check out QoS.  You can specify TOS tagging on your IAX
channels 
> in iax.conf, and the Polycom phones are able to respond to TOS tagging

> (in ipmid.cfg - or in the web interface under "Core Conf").  Maybe
they 
> are are trying to do two mutually exclusive kinds of TOS tagging?  You

> can tell the Polycom phone to just not respond to TOS.
> 
> - Noah
> 
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