[Asterisk-Users] (newby) IAX Trunk on low bandwidth connection

Thu Feb 2 20:44:49 MST 2006

hi,

I am just like yours a Satellite HUB operator that providing a voip. 
Before i am providing a ptp h323 with g723 codec boxes ranging 2 to 4 
port at 64kbps upstream and shared 2mbps downstream and now i come up to 
put asterisk using a g723 and g729 of digium but not work for me because 
when the remote end used data like email, browsing the voice suffer the 
quality. I look back on my last experiment and i see that SER and 
OPENSER is much better solutions to provide just a voice and voicemail 
to call out. The 64kbps with Qos in the remote config will help a lot 
better to provide a 99 % satisfaction to the customer.

//jollyr
Cosmin Prund wrote:

> At my HQ I’m instaling a 128kbit leased line connection, with 
> guaranteed bandwidth to the Internet; The telco promises less then 20 
> ms to the internet (to ronix.ro), no jitter and no packet loss. So I’m 
> hoping for 40 ms times to net and small jitter J This is my „hub”.
>
> For my „satelite” instalations I’m planning on grabing a connection 
> from a different provider (as this telco provider is expensive) but 
> I’m also considering a 64kbit leased line from the same provider, just 
> in case my VoIP doesn’t work with the cheeper providers. My remote 
> instalations will never have more then one „conversation” load, and 
> this conversation would be ZAP to IAX or SIP. That is, the distant 
> instalation will need to forward all calls coming in on the zap chanel 
> to my HQ Asterisk. That’s all it will ever do J. I’m not sure 
> „trunking” woud provide anything in this case as there will never be 
> more then one concurent conversation from the remote * to my HQ *. I’m 
> expecting IAX to provide better performance over SIP but not by much.
>
> Considering my remote * instalations will never have more then one 
> concurent conversation with my HQ and considering I can get a really 
> good 64kbit line I guess I’m OK. As for my HQ, I’m sure I’m OK because 
> I’ll get a 128 kbit line and I’ll be able to afford an upgrade to 
> 256kbit. I can actually go all the way to 2048 kbit, but that would no 
> longer be economically viable.
>
> So I’ll see how it goes, and I hope I’ll have the time to put in a 
> comment on the „low bandwidth” wiki on voip-info.org.
>
> Thanks to everyone for your help.
>
> ------------------------------------------------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *tim panton
> *Sent:* Thursday, February 02, 2006 11:05 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth 
> connection
>
> On 2 Feb 2006, at 08:09, Cosmin Prund wrote:
>
>
>
> Brrghhh: Bandwidth calculation is really foggy for me:
>
> Using the calculator I’m getting about 23 kbps for both incoming and 
> outgoing. What does this mean: Is a 64kbit link used at 71% capacity 
> ((23+23):64) or is it used at only 35% (23:64)? Will this vary over 
> time (i.e: does the codec generate more then average data at times? 
> How about less then average?)
>
> It depends on what sort of link you have. Most links are full duplex 
> (leased lines etc) which would be 35%
>
> but some radio based links are half duplex which would be 71%
>
> So for a 64k link you will (just about) get 3 729 calls.
>
> If all the calls between are between the same two servers, you can use 
> IAX trunking, which would push
>
> you up to 5 calls. (What that tells you is that for 729 and gsm, the 
> headers are as big as the data).
>
> You talk about satellite stations, if you are going for a hub and 
> spoke, you should put the hub
>
> on the highest bandwidth link.
>
>
>
> Thanks.
>
> ------------------------------------------------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of *Rob Lith
> *Sent:* Wednesday, February 01, 2006 11:40 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [Asterisk-Users] (newby) IAX Trunk on low bandwidth 
> connection
>
> What codec is that using. G.729 will give you 10 calls at best over 
> 256k unless you're trunking with IAX2? I don't know anyone using lpc10...
>
> Remember a G.729 8k codec turns into 23.63 Kbps with all the overheads...
>
> Regards
> Rob
>
> On 2/1/06, *Garth van Sittert* <garth at bitco.co.za 
> <mailto:garth at bitco.co.za>> wrote:
>
> Hi Cosmin
>
> You should be able to get about 12 simultaneous calls on a 128k line and
> about 28 on a 256k line according to asteriskguru's bandwidth calculator
> http://www.asteriskguru.com/tools/bandwidth_calculator.php.
>
> Kind Regards
> Garth
>
> BitCo Data Communications
> http://www.bitco.co.za
>
> Cosmin Prund wrote:
>> Hello everyone, this is my first post to the list, so hello again.
>>
>> We're a small company in Romania and we're trying to set up a really 
> small
>> version of "call center". That is, we want to get a few land-lines 
> from our
>> telco in different countys and "bridge" all calls to our HQ, in order to
>> make it cheeper for our clients to call us.
>>
>> Unfortunatelly there's no ISP in our area that can deliver a broadband
>> connection for anything less then an arm and a leg, so we're considering
>> runing an * <-> * connection using VoIP over a low bandwidth connection
>> (we're considering 128kbit but we might be able to go to 256kbit).
>>
>> The bandwidth price is not a problem for our "satelite" 
> installations, we
>> cand get acceptably priced broadband (~256kbit) so the distant *'s 
> will have
>> propper connections.
>>
>> My question:
>>
>> Is 128kbit a wide enough connection for 1 simultaneous conversation, 
> using
>> IAX protocol with the comercial version of the g729 codec?
>>
>> I'm expecting this to be engough for more then 1 conversation (after 
> all a
>> single line analog connection is rated at 64kbit and I'm getting 
> double that
>> bandwidth)
>>
>> Cosmin Prund
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com 
> <http://Easynews.com> --
>>
>> 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>
>> From - Wed
>>
>
> --
> Garth van Sittert
> BSc (Physics & Computer Science)
> -----------------
> Mobile: +27 (0)83 791 6662
> Email: garth at bitco.co.za <mailto:garth at bitco.co.za>
> Phone: 08600 BITCO
> Web: www.bitco.co.za <http://www.bitco.co.za>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com 
> <http://Easynews.com>--
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
>
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
>
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> http://www.westhawk.co.uk/
>
>------------------------------------------------------------------------
>
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>