BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp built right in, as well as other great apps, utilities, and security "auditing".
I suggest everyone have a copy in their arsenal, and it is free of course. Thanks, Steve Totaro On Fri, Oct 3, 2008 at 10:09 PM, Jai Rangi <[EMAIL PROTECTED]> wrote: > All, > > Just an update on this. This turned out to be a bug in Cisco firewall. We > ended up in upgrading the Firmware on the firewall. > > One thing I want to add, this was first time we used the fail over unit > during peak time. In the whole process (failover, upgrade and failover back > to active unit) was completely seamless. Did not had any down time, there > was just a pause for just 1 second in the audio. I was very impressed. > > -Jai > > > > On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi <[EMAIL PROTECTED]> wrote: > >> Oh yes, how could I forgot about that? >> Thank you, >> >> -Jai >> >> >> >> On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <[EMAIL PROTECTED]>wrote: >> >>> sipp can simulate RTP traffic. >>> >>> Jai Rangi wrote: >>> >>> > Al and Alex, >>> > Thank you for your input, >>> > Sorry TDM is not the option at this time :( . >>> > Everything has been great until last 2-3 days. Machine loads is not the >>> > issue, we have multiple asterisk server to share the load. Not much >>> > change in traffic. >>> > >>> > Now it been narrowed down to networking and we are trying to find out >>> > where the issue is? In our Firewall or SP's router. Does any one know >>> > of any tool to simulate RTP traffic. Its pain to find out the bad calls >>> > out of hundreds of calls. >>> > BTW, What should be right value for tos in sip.conf. >>> > We have >>> > tos=0x68 >>> > Dont remember how did I come up with this value. >>> > >>> > I found this >>> > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos >>> > >>> > tos=0x10 low delay >>> > tos=0x08 high throughput >>> > tos=0x04 high reliability >>> > tos=0x02 ECT bit set >>> > tos=0x01 CE bit set >>> > >>> > >>> > -Jai >>> > >>> > >>> > On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <[EMAIL PROTECTED] >>> > <mailto:[EMAIL PROTECTED]>> wrote: >>> > >>> > USE TDM Circuits - Voice Quality Good >>> > >>> > Alex Balashov wrote: >>> > > Jai Rangi wrote: >>> > > >>> > > >>> > >> All, >>> > >> >>> > >> I am having audio quality problem in some calls (1-2%) on >>> > asterisk. I >>> > >> captured RTP traffic using ethereal and this is what I found >>> > with the >>> > >> problematic calls. (Worst cases) >>> > >> Drop by Jitter buff: 25-75% >>> > >> Out of Seq: 50-100% (100 % means very very poor call quality). >>> > >> >>> > >> Has anyone had similar problem? If yes, can you please share >>> your >>> > >> experience on how did you fix this? >>> > >> >>> > > >>> > > Such poor performance is not fixable. The network, connectivity >>> > issues, >>> > > machine load, etc. needs to be addressed - the underlying cause, >>> in >>> > > other words. >>> > > >>> > > BTW, 100% out-of-sequence RTP packets leads to a lot more than >>> just >>> > > "very very poor call quality." I don't see how the conversation >>> > could >>> > > even be coherent in that situation. >>> > > >>> > > What is more likely is that Wireshark's RTP stats are giving you >>> some >>> > > distorted information. I've found its stream analysis to be >>> somewhat >>> > > buggy in that regard. >>> > > >>> > > >>> > >> I was wondering if I can decrease the MTU size to 250-500 on >>> the >>> > network >>> > >> card and use that card only for VoIP traffic. Will this have >>> any bad >>> > >> effect on sip traffic/packets? >>> > >> >>> > > >>> > > No. RTP packets are very small - much smaller than that MTU, or >>> any >>> > > reasonable MTU you could set. >>> > > >>> > > >>> > >>> > _______________________________________________ >>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com-- >>> > >>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> > Register Now: http://www.astricon.net >>> > >>> > asterisk-users mailing list >>> > To UNSUBSCRIBE or update options visit: >>> > http://lists.digium.com/mailman/listinfo/asterisk-users >>> > >>> > >>> > >>> > >>> ------------------------------------------------------------------------ >>> > >>> > _______________________________________________ >>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> > >>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> > Register Now: http://www.astricon.net >>> > >>> > asterisk-users mailing list >>> > To UNSUBSCRIBE or update options visit: >>> > http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> -- >>> Alex Balashov >>> Evariste Systems >>> Web : http://www.evaristesys.com/ >>> Tel : (+1) (678) 954-0670 >>> Direct : (+1) (678) 954-0671 >>> Mobile : (+1) (706) 338-8599 >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >>> Register Now: http://www.astricon.net >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype)
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users