Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection & their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the "top" command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will "skip" occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks.
On Sat, Nov 7, 2009 at 2:45 PM, John Timms <johngtimms at gmail.com> wrote:> I have a small-form-factor Asterisk server with an Intel Atom 230 CPU > (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu > Server 9.04 with the default Debian package manager installation of > Asterisk. (version 1.4)What kind of NIC are you using and what's the network config? ie Bellsouth -> router -> switch -> you Are you NAT'd? Where are your endpoints connected? (locally, outside?)> I have a very fast internet connection, so there is still plenty of > bandwidthwhat is the specs for "fast"?
John Timms wrote:> Hi. I'm having trouble figuring out why I'm not able to make many > concurrent VoIP calls on my system. I'm not aiming for a huge number, > because I have purposely bought a low powered system, but I would > think that I could get more. Here are the details: > > I have a small-form-factor Asterisk server with an Intel Atom 230 CPU > (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu > Server 9.04 with the default Debian package manager installation of > Asterisk. (version 1.4)Most of my installations are Soekris net5501's with 512MB ram and a 500mhz Geode LX processor. Unless Ubuntu is running a ton of extra junk in the background, that processor should be more than adequate.> Here is what is going on: I'm making outgoing calls (with .call files) > via SIP (using Vitelity's service, if anyone wants to know) with about > 55.0 ms latency between my Bellsouth DSL connection & their servers. > I'm using GSM-format prompts with GSM encoding (disallow=all, > allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. > I have a very fast internet connection, so there is still plenty of > bandwidth, and the "top" command shows that Asterisk is only at about > 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will > "skip" occcasionally, but cell phones have perfect quality.Your only connection to the PSTN is via SIP right? Then this is likely coincidental that 'landline' calls are different than 'cell phone' calls. The ONLY possibility is that the problem is with your SIP termination provider, but even that is unlikely. As Fred pointed out your DSL connection is likely the cause. Do you have any traffic shaping on the network? If not, you really should have a firewall that's capable of prioritizing voice traffic over bulk data traffic. What is the actual down and up speed of your DSL connection?> I don't think that 7 calls is very many, I'll be happy if I can get 10 > good-sounding calls. Can anyone give suggestions? (If this has been > hashed out elsewhere, I'm happy with a link to more information!)Use this calculator to see how much bandwidth 10 concurrent calls will take. http://www.asteriskguru.com/tools/bandwidth_calculator.php Darrick
If you've got a bellsouth dsl connection because of the way their system works even with doing qos on the link you can really only do about 8 calls before you start to run into problems with their setup. Tom -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Timms Sent: Saturday, November 07, 2009 2:45 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Help with concurrent VoIP calls Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu Server 9.04 with the default Debian package manager installation of Asterisk. (version 1.4) Here is what is going on: I'm making outgoing calls (with .call files) via SIP (using Vitelity's service, if anyone wants to know) with about 55.0 ms latency between my Bellsouth DSL connection & their servers. I'm using GSM-format prompts with GSM encoding (disallow=all, allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. I have a very fast internet connection, so there is still plenty of bandwidth, and the "top" command shows that Asterisk is only at about 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will "skip" occcasionally, but cell phones have perfect quality. I don't think that 7 calls is very many, I'll be happy if I can get 10 good-sounding calls. Can anyone give suggestions? (If this has been hashed out elsewhere, I'm happy with a link to more information!) Thanks. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Thanks for suggestions, everyone- I should have thought about jitter and latency as I began to use up more & more bandwidth. I was concerned that it was a problem with my configuration of Asterisk, but it looks like is really is a bandwidth issue. By the way, Joe- I've been in another situation with my cableco & Asterisk/VoIP (on a business connection!) and would frequently have trouble getting *one* call that sounded good, even though we had several megabits up & down, with no other traffic on the network. Charter's service is horrible- there were several times pinging Google took over 1 second. John Timms On Sat, Nov 7, 2009 at 2:45 PM, John Timms <johngtimms at gmail.com> wrote:> Hi. I'm having trouble figuring out why I'm not able to make many > concurrent VoIP calls on my system. I'm not aiming for a huge number, > because I have purposely bought a low powered system, but I would > think that I could get more. Here are the details: > > I have a small-form-factor Asterisk server with an Intel Atom 230 CPU > (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu > Server 9.04 with the default Debian package manager installation of > Asterisk. (version 1.4) > > Here is what is going on: I'm making outgoing calls (with .call files) > via SIP (using Vitelity's service, if anyone wants to know) with about > 55.0 ms latency between my Bellsouth DSL connection & their servers. > I'm using GSM-format prompts with GSM encoding (disallow=all, > allow=gsm in sip.conf) and I'm able to make about 7 concurrent calls. > I have a very fast internet connection, so there is still plenty of > bandwidth, and the "top" command shows that Asterisk is only at about > 5% CPU and 10% RAM. Even with only 7 calls, a landline phone will > "skip" occcasionally, but cell phones have perfect quality. > > I don't think that 7 calls is very many, I'll be happy if I can get 10 > good-sounding calls. Can anyone give suggestions? (If this has been > hashed out elsewhere, I'm happy with a link to more information!) > > Thanks. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091109/a6a6e8cf/attachment.htm