I understand. I'm interested in simulate the real situation because I'm doing an academic comparative between algorithms, and is really interesting have all possible situations. In the real situation I use a E1 to connect a PBX through a R2 link, so I want to try change DAHDI to 10 ms... What should I modify? Any other sugestion to make it works?
2011/9/14 Kevin P. Fleming <[email protected]> > On 09/14/2011 02:37 PM, Gustavo Santos wrote: > >> I'm trying to simulate the situation: >> >> SIP <----> Asterisk <-------> PSTN >> >> In this case 16 ms works? >> >> I've read in voip-info: "Simplistically, you'd need a "tail circuit" >> (the distance between your echo canceller and the source of the echo) of >> over 2500 miles to acheive an echo path of 30ms [...] Asterisk's default >> of 128taps will therefore handle echo paths of up to 16ms, and is >> therefore probably good for most things.". >> > > You are missing some basic details of the environment here. I'll try to > explain. > > In the diagram you've shown above (assuming there is an FXO port in the > Asterisk server connected to an FXO line from the PSTN), there are > potentially two sources of line echo: the 2/4 wire hybrid in the FXO port, > and the 2/4 wire hybrid at the far end of the FXO line (and potentially even > farther into the PSTN, but we can ignore that here). Echo caused by the FXO > port hybrid would be heard by the person at the other end of the FXO line > (across the PSTN), and would not be cancelled by any echo canceller on the > FXO card or in DAHDI. Echo caused by the far end would be heard by the user > of the SIP phone, and could potentially be cancelled by an echo canceller on > the FXO card or in DAHDI. > > That quote you've included above is correct: assuming a *TRADITIONAL* PSTN > link (no VoIP, no packetization of audio, all circuits either analog or > TDM), the echo generated by the far end of the FXO line will likely not be > more than 16ms after the transmission. In this case, a 16ms echo canceller > window will be adequate. If an echo (primary or secondary) is generated by > the real "far end" (across the PSTN), it could easily be delayed by 30ms (or > much more). In these cases, having a 64ms or 128ms echo canceller window is > beneficial, and with modern hardware is not expensive to provide (or harmful > in any way). > > However... using Asterisk with an FXS card and the Echo() application is > *NOT* a 'PSTN simulator'. When an audio signal is received into the FXS > card, it will take 20-40ms to be sent back out the FXS card, depending on > packetization intervals, scheduling delays and other factors. This is > because, as I stated previously, Asterisk is internally a 'voice over > packet' system, and it does not have any way to forward audio in anything > less than reasonable size packets. For cards driven by DAHDI, 'reasonable' > defaults to 20ms, although it could be changed to 10ms with a corresponding > increase in CPU overhead... but even if you did change it, it is likely that > under many situations the echoed audio would be delayed by more than 16ms. > > If you *need* to test an echo canceller configured with a tiny 16ms window, > you'll have to find another way of generating echo for it to be tested > against. > > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- Atenciosamente, Gustavo Santos.
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