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

[Paul Fielding]

----- Original Message ----- 
> 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.

I'd still be wondering if there's something else.  I, too, experience choppy 
SIP connectivity from external IPs, but as I've mentioned in previous 
postings, I have a Vonage ATA that seems to have no problems keeping a 
crystal clear connection as it leaves my place and goes to Vonage's servers, 
so I think there must be more to it than QoS.   I have to believe that 
there's some more jitter correction or other such buffering that could 
berhaps be played with, though I don't know what it would be.... 
*shrug*.....?

regards,

Paul


>
>
> 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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