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 <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] 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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