----- Original Message ----- From: "Lonnie Abelbeck" <[EMAIL PROTECTED]> To: "Discussion of AstLinux - Asterisk on Compact Flash" <[EMAIL PROTECTED]> Sent: Tuesday, February 28, 2006 8:16 AM Subject: Re: [Astlinux-users] Possible for 50 SIP ConcurrentCalls? Soekris/WASP
[...] > I'm just saying with canreinvite=no, SIP vs IAX2, the Net4801 has more > work to do trunking via IAX2 vs SIP. > > Maybe the CPU horsepower difference is very small, but combining all > the voice streams into one stream, in a timely manner, doesn't seem > like a small thing to me. One is not forced to turn trunking on, and if that is done there are significant bandwidth savings over SIP+RTP (see http://www.asteriskguru.com/bandwidth_calculator.php , or the tables at http://www.convergence.com.pk/iax2/trunked.html ). > For VoIP service providers, IAX2 does not scale, but SIP can scale > nicely. Maybe there are lessons that apply to the embedded systems as > well. > > My VoIP provider, teliax.com, has IAX in their name, but publicly admit > that the voice quality is better if asterisk users trunk via SIP > instead if IAX2. Teliax's users agree with this recommendation, as > much as we like IAX2. Things should improve with the latest multithreaded code by Mark Spencer: http://lists.digium.com/pipermail/asterisk-dev/2006-February/019002.html I'm also curious to know if anybody has tried the IAX implementation in Opal (http://www.voxgratia.org/docs/derek/ ). That was said to be less timing-sensitive than the traditional libiax2 used in Asterisk until now. Enzo _______________________________________________ Astlinux-users mailing list [email protected] http://lists.kriscompanies.com/mailman/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to [EMAIL PROTECTED]
