Hi, We have two Asterisk servers connected over IAX, with very limited bandwidth 256Kbs. When we make calls between these two Asterisk servers the sound is very choppy, no matter whether we use jitter buffer or not. However, when we make calls using Skype, the sound is perfect. Can anyone help us troubleshoot this IAX issue that we are experiencing? Best regards, Stojan Sljivic
I have 3 POTS lines that I want to use with Asterisk, I am looking at prices for FXO cards and the cards with echo cancellation are really pricey... is echo cancellation really worth it for a 3 or 4 line system? Will I notice a difference without the echo cancellation? Thanks Keith Schmidt
On Mar 14, 2006, at 9:04 AM, Keith Schmidt wrote:> I have 3 POTS lines that I want to use with Asterisk, I am looking at > prices for FXO cards and the cards with echo cancellation are really > pricey... is echo cancellation really worth it for a 3 or 4 line > system? Will I notice a difference without the echo cancellation?This depends greatly on the quality of your PSTN line and the distance from the CO (central office). In my case, with a two wire loop over 15000 feet, I definitely had echo issues that made cheapo FXO unusable. Although with a Digium card, you also have the option of using software based echo cancellation. I have no experience with that. Good echo cancellation is worth it in my opinion. Marty
On 14 Mar 2006, at 16:44, Stojan Sljivic - GDS wrote:> Hi, > > We have two Asterisk servers connected over IAX, with very limited > bandwidth > 256Kbs. > When we make calls between these two Asterisk servers the sound is very > choppy, no matter whether we use jitter buffer or not. > > However, when we make calls using Skype, the sound is perfect. > > Can anyone help us troubleshoot this IAX issue that we are > experiencing?Maybe if you tell us some more :-) What codecs are you using? Are you using Trunked IAX? How many calls at a time? What is the ping time between the systems? Any error messages ?
Hi,> Are you using Trunked IAX?Currently we do not use trunking.> How many calls at a time?All the test we have performed so far were with only one active call.> What codecs are you using?We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 are in the play.> What is the ping time between the systems?Ping stats are: Server 1: 50 packets transmitted, 50 received, 0% packet loss, time 49491ms rtt min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms Server 2: 50 packets transmitted, 49 received, 2% packet loss, time 49523ms rtt min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms> Any error messages ?There are no error messages in the console. Regards, Stojan Sljivic> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Tim Panton > Sent: Tuesday, March 14, 2006 19:10 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] IAX choppy sound > > > > On 14 Mar 2006, at 16:44, Stojan Sljivic - GDS wrote: > > > Hi, > > > > We have two Asterisk servers connected over IAX, with very limited > > bandwidth > > 256Kbs. > > When we make calls between these two Asterisk servers the > sound is very > > choppy, no matter whether we use jitter buffer or not. > > > > However, when we make calls using Skype, the sound is perfect. > > > > Can anyone help us troubleshoot this IAX issue that we are > > experiencing? > > Maybe if you tell us some more :-) > What codecs are you using? > Are you using Trunked IAX? > How many calls at a time? > What is the ping time between the systems? > Any error messages ? > > _______________________________________________ > --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 >
On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote:> Hi, > >> Are you using Trunked IAX? > Currently we do not use trunking. > >> How many calls at a time? > All the test we have performed so far were with only one active call. > >> What codecs are you using? > We have set the bandwith=low, so I think that G.723.1, GSM, and LPC10 > are in > the play.If you have 256kbits/s available and want to make a maximum of 2 calls you could try something using ulaw (~80kbits/s) anyhow, I would explicitly set the codec so that you can compare them. eg: disallow=all allow=ulaw> >> What is the ping time between the systems? > Ping stats are: > Server 1: > 50 packets transmitted, 50 received, 0% packet loss, time 49491ms > rtt min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms> Server 2: > 50 packets transmitted, 49 received, 2% packet loss, time 49523ms > rtt min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms >That is quite a variation, over an already longish ping time. you probably need to do some traffic shaping at your routers to give IAX priority. If you are getting good results from skype over the same link, you could try examining the TOS bits in the skype packets and setting the IAX to use the same TOS bits since that may be what is making the difference.>> Any error messages ? > There are no error messages in the console.Just to check, can you get decent call quality between 2 IAX clients on the same (local server)?> > Regards, > Stojan Sljivic >Hope that helps Tim
Hi Tim,> Just to check, can you get decent call quality between 2 IAX > clients on > the same > (local server)?I have never tested that since we have no IAX phones. We use SIP phones and IAX is used for connecting two Asterisk servers. Regards, Stojan Sljivic> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Tim Panton > Sent: Wednesday, March 15, 2006 13:08 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] IAX choppy sound > > > > On 15 Mar 2006, at 09:02, Stojan Sljivic - GDS wrote: > > > Hi, > > > >> Are you using Trunked IAX? > > Currently we do not use trunking. > > > >> How many calls at a time? > > All the test we have performed so far were with only one > active call. > > > >> What codecs are you using? > > We have set the bandwith=low, so I think that G.723.1, GSM, > and LPC10 > > are in > > the play. > > If you have 256kbits/s available and want to make a maximum > of 2 calls you could try something using ulaw (~80kbits/s) > anyhow, I would explicitly set the codec so that you can compare them. > eg: > > disallow=all > allow=ulaw > > > > > >> What is the ping time between the systems? > > Ping stats are: > > Server 1: > > 50 packets transmitted, 50 received, 0% packet loss, time > 49491ms rtt > > min/avg/max/mdev = 190.198/213.028/283.275/25.307 ms > > > > Server 2: > > 50 packets transmitted, 49 received, 2% packet loss, time > 49523ms rtt > > min/avg/max/mdev = 190.089/214.855/544.880/59.052 ms > > > > That is quite a variation, over an already longish ping time. > you probably need to do some traffic shaping at your routers > to give IAX priority. If you are getting good results from > skype over the same link, you could try examining the TOS > bits in the skype packets and setting the IAX to use the same > TOS bits since that may be what is making the difference. > > >> Any error messages ? > > There are no error messages in the console. > > Just to check, can you get decent call quality between 2 IAX > clients on > the same > (local server)? > > > > > Regards, > > Stojan Sljivic > > > > Hope that helps > > Tim > > _______________________________________________ > --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 >