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

[Gabriel Afana]

What options are there at all with Asterisk that I can play with to *tweak* 
it to try to optimize sound?  I've already check and played with the QoS 
(0x10, 0x18..etc)...no difference.

Gabe


----- Original Message ----- 
From: "Paul Fielding" <paul.fielding at shaw.ca>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Thursday, January 20, 2005 1:18 PM
Subject: Re: [Asterisk-Users] top-notch servers/OS/network, ulaw codec - 
soundstill choppy


> ----- 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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>>
>>
>> -- 
>> Is it something someone said, was it something someone said?
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