[Asterisk-Users] VoIP Provider problems

Michael D Schelin mike at shelcomm.com
Wed Mar 30 19:55:14 MST 2005


Give me a try!  www.shelltel.com  And don't use G711 for your calls. 
invest in the G729 codec. you'll find your calls will start working 
better.  I'm a G729 shop.
Thanks
Michael D. Schelin
626-814-2454

Max W Blackmer Jr wrote:

>>We recently configure an asterisk server to use with an VoIP provider
>>to make calls to a PSTN. We use (voipjet, nufone, diamond....)
>>
>>We feel that we haven't got the quality that we hope. Sometimes our
>>calls gets mute, or we feel communication cuts on our phone calls.
>>We have got an QOS router (Draytek) reserving 1/2 of our wideband to
>>the SIP an IAX2 protocols, and an ADSL line about 2 Mb.
>>    
>>
>
>ADSL has slower upload speeds than download speeds (your 2Mbps is
>download). so you may have problems with your outgoing packets of
>sound. g.711 codec (the default codec for most voip providers because
>there is virtually no sound quality loss) uses about 84Kbps per channel
>or simultaneous connection. For example if you have an Upload speed of
>128Kbps. and you try to have 2 phone conversations you would need
>168kbps transfer speed. That is 40kbps more than your upload speed.
>This is a major problem with ADSL the upload and download speeds are
>not equal.
>
>Another potential problem is that your provider is over subscribed for
>the available bandwidth. What this means is that when allot of people
>are using their connection to your provider. The provider may not be
>able to handle all those users at once and packets get dropped or
>delayed. Dropping or delaying packets is very bad for VoIP especially
>if they do not do QoS or ToS routing which most providers do not.
>
>What is your upload speed?
>
>Some other possibilities are to use some compression codecs which will
>cause some sound quality loss like gsm or  iLibc and g.729 to pack more
>calls in the limited bandwidth limitations.  Another option is to use
>SDSL where the speeds of  both the upload and download are the same.
>
>  
>
>>We feel our quality decrease when in US are about 9:00 or 10:00 in the morning.
>>    
>>
>
>This time is when businesses in the us are opening and starting to do
>business In the united states. Both for phones and Data.
>
>
>  
>
>>We do not know if this is it correct or all the people using VoIp
>>provider feel the same quality?
>>    
>>
>
>This may mostly be in relation to you Internet provider and how many
>hops you have to take to get to the VoIP provider and if they
>oversubscribe their bandwidth capacity. One provider may be good for
>one person with one person in a different  ISP than an ISP you have.
>And you are even right next door to each other. This is as a result of
>how the internet is connected and may not nessessarly be geographic.
>For example you may be connecting to a server in your own city lets say
>Chicago but you are actually routed to San Francisco then back to
>Chicago. But it will not always take the same path the next time you
>may be routed through New York. This is a simplification of how it
>works.  The closer you are to a Tier 1 provider(they own the major
>trunks interconnects) the less time it will take to get to your target.
>
>  
>
>>Anyone knows any provider without this kind of problems?
>>    
>>
>
>I have seen many Providers have both Good and bad connection links. It
>is best to have a provider that routes with QoS and/or ToS within their
>routers and have only one or two hops between your provider and a tear 1
>provider.
>
>  
>
>>Witch provider do you use to get the best sounds quality?
>>    
>>
>
>It is not that simple. But you can begin by doing a traceroute to the
>many providers at different times of the day. This will see the route
>changes and time delays between hops to get to VoIP Providers gateways.
>
>Hope this helps in understanding the problems involved with choosing a
>provider.
>
>Thanks,
>
>Max
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>  
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050330/279e0c7f/attachment.htm


More information about the asterisk-users mailing list