One-way audio quality problems (beginner user)


#1

First of all, scenario:

Asterisk on PIV 1.8GHz with 768MB of RAM on an IDE hard drive, on a 512Kbps symetric cable connection with no packet filtering whatsoever in front, with a public IP address. Asterisk 1.2.4.

Zap 1/1 is a Wildcard X100P.

Client is a X-Lite softphone running on my laptop, an Asus A4L with Mobile PIV 3.06Ghz with 512MB RAM, behind a NATting gateway on a 20Mbps connection.

SIP configuration is as follows:

[101]
host=dynamic
type=friend
secret=xxxxxx
context=plataforma
callerid=Leonardo Murillo
nat=yes
canreinvite=yes
mailbox=101@plataforma
allow=gsm
allow=ulaw
dtmfmode=rfc2833

Latency between client and server average 20ms with low 10ms and high 40ms.

The problem is that the audio that is outgoing, meaning, from Client, to Asterisk, to PSTN is terrible :imp: , breaks up, crackles, pops, hisses, spits at you when youre not looking :stuck_out_tongue:, but the incoming audio, from PSTN, to Asterisk over the internet to my sip client is perfect.

The problem does not happen when the same laptop and client are connected to the same network as the asterisk box.

My guess is, it’s either gotta be something to do with NAT, something to do with bandwidth though I doubt that (the 512Kbps connection is completely idle except for that one phone call and my SSH session), or else Ive got no idea.

I’ve done a search on the forum and even though I’m sure this question must have been asked a million times, couldnt find much information to guide me towards the possible solution.

All advice will be greatly appreciated.