[Asterisk-Users] asteisk, sip & NAT

[Andrew Radke]

Hervé Thibaud wrote:

>Le dim 22/06/2003 à 16:18, Hervé Thibaud a écrit :
>...
>  
>
>>I try to connect directly the both to fwd.pulver.com and now i have a
>>perfect sound but the question is perhaps links after opening session 
>>is only on the local networks with 10Mb/s.
>>Once i can (when i'll have an external user to call) i'll try.
>>    
>>
>
>It's like i thought, the sound is nasty with many blanks when i try a
>connection on internet and ISDN bandwith on one channel 64kb/s is not
>enough and I cannot have ADSL here. I saw Sjphone use codec 711 only and
>use a bandwith of 64kb/s so. 
>Is anybody that has a free or a very cheap solution (it's to try
>asterisk) to have an IP phone hardware or software with G.723.1 codec
>
>Other thing, I would like to try X-Lite but i have pb with registration,
>i don't know how to write settings, i have an error (for example) : 
>File chan_sip.c, Line 4412 (handle_request) : Registration from 'roseau
><sip:roseau at 192.168,0,1>" failed for '192,168,0,1'
>I try many form of settings but didn't succeed. 
>
X-Lite support speex (SPX on the X-Lite screen). This is a very 
impressive codec that will even allow you to talk with someone over a 
_modem_ with a little bandwidth to spare for other stuff. Obviously the 
modem adds a degree of latency but it's still impressive. So with the 
latest technology, etc we've managed to get a _voice_ conversation to 
travel over a standard phone line. ;-) But seriously, it is impressive.

Regards,

Andrew Radke                                       ,-_|\
andrew at radke.iig.com.au   mobile: +61 412 798593  /     \
Member, System Administrators Guild of Australia  \_,-._*
------------------------------------------------       o
"I didn't know it was impossible when I did it."