[Asterisk-Users] top-notch servers/OS/network, ulaw codec - sound still choppy

[Gabriel Afana]

Great, thanks for the info.  This is a service provided from my colo, so I 
will have to give them a call and find out whats up with their router 
settings.  As for packet loss, how do I check for that?

Gabe

----- Original Message ----- 
From: "Jon Radon" <jonr800 at gmail.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, January 20, 2005 12:32 PM
Subject: Re: [Asterisk-Users] top-notch servers/OS/network,ulaw codec - 
sound still choppy


>I see the sip user is an external ip.  I would take a look at your QoS
> settings on your router.  Make sure the voice traffic is getting the
> priority it deserves.  Also, check for packet loss.
>
>
> On Thu, 20 Jan 2005 11:26:02 -0800, Gabriel Afana <adtomi at adelphia.net> 
> wrote:
>> Hi,
>>    My SIP calls are sounding a little choppy.  I've did my research but
>> everything looks right on my end...what am I missing?
>>
>> Running RedHat ES 3.0 on dual AMD Opteron servers.  My system is 
>> cololocated
>> in downtown LA and is fed via a gigabit handoff from XO, AT&T, Level 3 
>> and
>> Wiltel (I have a 100Mb didicated line).  So I dont think its the Servers,
>> its the network, Asterisk is working fine and all codecs look 
>> right...what
>> could be the cause?
>>
>> **SNIP FROM SIP.CONF***
>> [general]
>> context=default                 ; Default context for incoming calls
>> port=5060                       ; UDP Port to bind to (SIP standard port 
>> is
>> 5060)
>> ;bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds 
>> to
>> all)
>> srvlookup=yes                   ; Enable DNS SRV lookups on outbound 
>> calls
>>                                ; Note: Asterisk only uses the first host
>>                                ; in SRV records
>>
>> allow=ulaw                      ; Allow codecs in order of preference
>> *************************
>>
>> ga0*CLI> sip show channels
>> Peer             User/ANR    Call ID      Seq (Tx/Rx)   Format
>> 64.201.99.247    9092479878  2fd496bf330  00103/00105   ulaw
>>
>> ga0*CLI> show version
>> Asterisk 1.0.3 built by root at g0.gafana.com on a x86_64 running Linux
>>
>> P.S.  in my sip.conf file, it looks like I am only allowing the ulaw
>> codec...could that cause a problem if I happen to need to call somebody 
>> that
>> doesn't support ulaw?
>>
>> Gabe
>>
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>
>
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