nr k
2005-Nov-08 22:00 UTC
[Asterisk-Users] maximum concurrent conference peers in asterisk
Hi all I have a 2 mbps bandwidth in my central location and 64 kbps in all my branch office. what is the maximum concurrent conference peers possible through asterisk.p placed the asterisk in my central location and i have ip phones in all my branch i am using g729 codec(for testing). regards ramakrishnan.n __________________________________ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com
Matt Riddell
2005-Nov-08 22:17 UTC
[Asterisk-Users] maximum concurrent conference peers in asterisk
nr k wrote:> Hi all > > I have a 2 mbps bandwidth in my central location and > 64 kbps in all my branch office. what is the maximum > concurrent conference peers possible through > asterisk.p placed the asterisk in my central location > and i have ip phones in all my branch i am using g729 > codec(for testing).is the 64 kilo bits or kilo bytes? -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
trixter aka Bret McDanel
2005-Nov-09 02:16 UTC
[Asterisk-Users] maximum concurrent conference peers in asterisk
On Wed, 2005-11-09 at 21:32 +1300, Matt Riddell wrote:> nr k wrote: > > hi generally we describe the bandwidth in kilobits per > > second only. > > Cool, just checking, it seemed pretty low. > > According to http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 you should > be able to do 4 calls with g729. >Bandwidth is a tricky issue. You have your IP + UDP + RTP + whatever headers (iax2 combines stuff so potentially that skews this a bit) but something most often forgotten is link layer framing. Take ATM (DSL uses ATM as do many other links). ATM transmitted data is chopped up into 53 byte cells. Each cell has a 5 byte header. This leaves 48 bytes for payload per ATM cell. Lets say your total packet IP header on down is 80 bytes. This means that on the ATM layer you have 2 ATM cells with 16 bytes of padding. This is really only 83% efficient network wise. This is per RTP packet. By adjusting your sample size you can try to fill the cell completly so you dont waste extra bandwidth on padding (ATM cells can contain no more than 1 packet and they are padded to fill the cell. so every 48 bytes of payload is another cell). You dont want your sample size too small however because that causes more IP overhead, too large and it can degrade call quality (imagine a 30ms jitter buffer with 30ms sample sizes, that means only 1 packet goes in the jitter buffer, with only one packet you have the effect of no buffer at all, reordering packets is impossible, delayed packets cant be normalized timewise, etc). Its a really fine balance and something you should consider if you really want to tune your VoIP to your network. Obviously once its handed off to another network it becomes hard to create packets tuned for a network you dont control, but its fair to assume that the majority of backbone providers are doing ATM so by tweaking this you may find that your voice traffic works better over the net at large too ... YMMV I didnt pay attention to what type of link the 64Kbps links were (I dont think it was specified initially) so I dont know what framing is used, but this is something to consider. By not paying attention to this fine detail you can waste a lot of bandwidth then wonder why you start to have lossy performance when raw bandwidth meters suggest you shouldnt have any loss. This was something that I presented to the Sacramento Asterisk Users Group last friday, although my power point presentation doesnt give the subject the coverage it needs, most of that was audible. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20051109/62af04ff/attachment.pgp