http://bugs.digium.com/bug_view_page.php?bug_id=0000149
bkw
On Tue, 2 Sep 2003, Eduardo Goncalves wrote:
On Tue, 2 Sep 2003 10:27:53 -0500 (CDT) Brian West <[EMAIL PROTECTED]> wrote:
> I opened a request on bugs.digium.com for this feature. The 6k and 8k > codecs are very impressive also. >> > bkw >Where can I see the status of this request?> Eduardo
[]'s
Spoiler: Read to the bottom about how to get 400 calls into a megabit. Maybe.
I'm ashamed to say I had not actually looked at the lower-bandwidth encoding options of Speex in the past, and skipped right over that section of the text in favor of the high bandwidth bitrates. What a mistake! I am extremely impressed with the 4kbps VBR rate for Speex, at least from the samples on the Speex website.
If the sound quality is as good as advertised at the low bitrates, the addition of selectable features for Speex would truly be an asset to Asterisk on a per-call basis (heck, even just per-peer.) I have several clients who need to move traffic across international IP capacity, and the low-bandwidth option of choice to them is G.729 (LPC10 is not an option due to sound quality issues.) The very interesting features of VAD and VBR look to be (on paper, at least) a real win as well, with the channel bitrate being reduced even further by silence and sound complexity compression.
Exposing codec feature selections to the dialplan would be interesting, but I expect Mark will want to (correctly) implement a generic method for doing this. However, are there any other codecs that Asterisk supports that have the ability to use different options (bitrate, VBR, VAD)? Is it worthwhile to make this a generic function of some sort, or is it sufficient to make specific techniques just for Speex? (${SPEEX-BITRATE}, or ${SPEEX-VBR} to give crude examples.)
Why am I so excited about this? The point of VoIP for most of my customers is twofold: the first point is the addition of new and novel services that they would not be able to offer previously without investing a lot in hardware. The second (and for some, the primary) point is that VoIP allows the transmission of voice packets over a less expensive packetized path than TDM. Thus, the biggest number on their minds is "Cost of Bandwidth!" The more voice streams you can pack into the bandwidth, the less they pay for bandwidth, and thus the larger the profit margin - very simple equation. So, they're really REALLY interested in any way to get more calls into the same number of bits per second, and Speex seems to have some interesting options in that arena.
Combined with the clever use of trunking with IAX2, I could possibly see (looking at back-of-napkin, totally theoretical numbers) something like 400 calls in a megabit between two Asterisk servers. That number seems wrong to me, and I expect my first impression is correct, but here's the math: with my IAX2 tests which I documented previously on this list, I got a theoretical 103 calls into a megabit of bandwidth with G.729 at 9.6kbps per additional call. Now, the Speex codec can be turned down to 4kbps, so I can get 2.4 Speex calls into the same space that I fit one G.729 call. So, (2.4 * 103 = 247) into a megabit. Now, usually only one person is talking at a time. This means VAD would be active on 50% of the channels, thus eliminating traffic in one way for all calls. I'm sure that's not quite accurate due to background noise and overtalk, so let's say that only 30% of the legs are empty at any one time due to VAD. So, that's an additional few channels, so now we're at (247 + (247 * .3) = 321) total channels. Now I move into the really unstable math (i.e.: I'm making this up based on wild fantasy.) If VBR is implemented, maybe/hopefully/possibly that permits us another 25% savings on bits per second, so that turns into (321 + (321 * .25) = 401) channels in a single megabit. This seems impossible. Anyone care to shoot holes in these numbers?
JT
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