David, I have IAX trunks running between the US and S. America using the GSM codec and jitterbuffer=yes and the quality seems very good to my ears. Don't have the details of the jitterbuffer parameters right now but hopefully this will give you some useful feedback. Good luck. On Thu, 2005-09-08 at 16:49 -0400, David Hajek wrote:> Hi- > > I'm using IAX between two boxes, where one box is located in US and the > second in Europe. I'm trying to achieve the best voice quality and > mainly reliability between these boxes and looking for hints and > experience of others. > > Facts: > - Asterisk 1.0.7 > - RTT varies from 130-170 ms, depends on time and actual Internet > throughput > > Questions: > - What is the sugested codec for such setup? Now I'm using ULAW, but > realizing it may not be the best choice. Sometimes I can hear broken > audio. Maybe speex is better choice? > - Jitter buffer, yes/no? What are the suggested values. Currently I'm > using these values: > jitterbuffer=yes > dropcount=10 > maxjitterbuffer=500 > maxexcessbuffer=300 > minexcessbuffer=20 > jittershrinkrate=2 > - Trunking? Is it reliable enough? >
Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David
Pay the license fee and get the GSM codec would probably be best. The fee is nominal and the codec is a good one... $0.02 W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 1:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice over atlantic Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup? Now I'm using ULAW, but realizing it may not be the best choice. Sometimes I can hear broken audio. Maybe speex is better choice? - Jitter buffer, yes/no? What are the suggested values. Currently I'm using these values: jitterbuffer=yes dropcount=10 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=20 jittershrinkrate=2 - Trunking? Is it reliable enough? Thanks for any hints. -- David _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
Nice. Thanks. What Asterisk version? Can you lookup jitterbuffer settings? Thanks a lot.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > asterisk groups > Sent: Thursday, September 08, 2005 7:22 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] voice over atlantic > > David, > I have IAX trunks running between the US and S. America using > the GSM codec and jitterbuffer=yes and the quality seems very > good to my ears. > Don't have the details of the jitterbuffer parameters right > now but hopefully this will give you some useful feedback. > > Good luck. > > > On Thu, 2005-09-08 at 16:49 -0400, David Hajek wrote: > > Hi- > > > > I'm using IAX between two boxes, where one box is located in US and > > the second in Europe. I'm trying to achieve the best voice > quality and > > mainly reliability between these boxes and looking for hints and > > experience of others. > > > > Facts: > > - Asterisk 1.0.7 > > - RTT varies from 130-170 ms, depends on time and actual Internet > > throughput > > > > Questions: > > - What is the sugested codec for such setup? Now I'm using > ULAW, but > > realizing it may not be the best choice. Sometimes I can > hear broken > > audio. Maybe speex is better choice? > > - Jitter buffer, yes/no? What are the suggested values. > Currently I'm > > using these values: > > jitterbuffer=yes > > dropcount=10 > > maxjitterbuffer=500 > > maxexcessbuffer=300 > > minexcessbuffer=20 > > jittershrinkrate=2 > > - Trunking? Is it reliable enough? > > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >
Probably missing something here. Never heard of GSM commercial licence for asterisk. Do you have any URLs? Thanks.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Wiley Siler > Sent: Thursday, September 08, 2005 11:09 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > Pay the license fee and get the GSM codec would probably be best. > The fee is nominal and the codec is a good one... > $0.02 > > W > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > David Hajek > Sent: Thursday, September 08, 2005 1:50 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] voice over atlantic > > Hi- > > I'm using IAX between two boxes, where one box is located in > US and the second in Europe. I'm trying to achieve the best > voice quality and mainly reliability between these boxes and > looking for hints and experience of others. > > Facts: > - Asterisk 1.0.7 > - RTT varies from 130-170 ms, depends on time and actual > Internet throughput > > Questions: > - What is the sugested codec for such setup? Now I'm using > ULAW, but realizing it may not be the best choice. Sometimes > I can hear broken audio. Maybe speex is better choice? > - Jitter buffer, yes/no? What are the suggested values. > Currently I'm using these values: > jitterbuffer=yes > dropcount=10 > maxjitterbuffer=500 > maxexcessbuffer=300 > minexcessbuffer=20 > jittershrinkrate=2 > - Trunking? Is it reliable enough? > > Thanks for any hints. > > -- > David > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >
http://www.digium.com/index.php?menu=product_detail&category=extras&prod uct=G729 -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Probably missing something here. Never heard of GSM commercial licence for asterisk. Do you have any URLs? Thanks.> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley > Siler > Sent: Thursday, September 08, 2005 11:09 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > Pay the license fee and get the GSM codec would probably be best. > The fee is nominal and the codec is a good one... > $0.02 > > W > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David > Hajek > Sent: Thursday, September 08, 2005 1:50 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] voice over atlantic > > Hi- > > I'm using IAX between two boxes, where one box is located in US and > the second in Europe. I'm trying to achieve the best voice quality and> mainly reliability between these boxes and looking for hints and > experience of others. > > Facts: > - Asterisk 1.0.7 > - RTT varies from 130-170 ms, depends on time and actual Internet > throughput > > Questions: > - What is the sugested codec for such setup? Now I'm using ULAW, but > realizing it may not be the best choice. Sometimes I can hear broken > audio. Maybe speex is better choice? > - Jitter buffer, yes/no? What are the suggested values. > Currently I'm using these values: > jitterbuffer=yes > dropcount=10 > maxjitterbuffer=500 > maxexcessbuffer=300 > minexcessbuffer=20 > jittershrinkrate=2 > - Trunking? Is it reliable enough? > > Thanks for any hints. > > -- > David > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
Yep. Thats G729, not GSM. Btw, GSM codec implemented in Asterisk is EFR?> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Wiley Siler > Sent: Friday, September 09, 2005 12:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > > http://www.digium.com/index.php?menu=product_detail&category=e > xtras&prod > uct=G729 > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > David Hajek > Sent: Thursday, September 08, 2005 2:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > Probably missing something here. Never heard of GSM > commercial licence for asterisk. > > Do you have any URLs? > > Thanks. > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley > > Siler > > Sent: Thursday, September 08, 2005 11:09 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] voice over atlantic > > > > Pay the license fee and get the GSM codec would probably be best. > > The fee is nominal and the codec is a good one... > > $0.02 > > > > W > > > > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David > > Hajek > > Sent: Thursday, September 08, 2005 1:50 PM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] voice over atlantic > > > > Hi- > > > > I'm using IAX between two boxes, where one box is located in US and > > the second in Europe. I'm trying to achieve the best voice > quality and > > > mainly reliability between these boxes and looking for hints and > > experience of others. > > > > Facts: > > - Asterisk 1.0.7 > > - RTT varies from 130-170 ms, depends on time and actual Internet > > throughput > > > > Questions: > > - What is the sugested codec for such setup? Now I'm using > ULAW, but > > realizing it may not be the best choice. Sometimes I can > hear broken > > audio. Maybe speex is better choice? > > - Jitter buffer, yes/no? What are the suggested values. > > Currently I'm using these values: > > jitterbuffer=yes > > dropcount=10 > > maxjitterbuffer=500 > > maxexcessbuffer=300 > > minexcessbuffer=20 > > jittershrinkrate=2 > > - Trunking? Is it reliable enough? > > > > Thanks for any hints. > > > > -- > > David > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@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 > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@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 > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >
Ooops... meant G729 but seems like other suggestion of GSM might do... W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Hajek Sent: Thursday, September 08, 2005 3:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] voice over atlantic Yep. Thats G729, not GSM. Btw, GSM codec implemented in Asterisk is EFR?> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley > Siler > Sent: Friday, September 09, 2005 12:08 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > > http://www.digium.com/index.php?menu=product_detail&category=e > xtras&prod > uct=G729 > > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David > Hajek > Sent: Thursday, September 08, 2005 2:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > Probably missing something here. Never heard of GSM commercial licence> for asterisk. > > Do you have any URLs? > > Thanks. > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Wiley > > Siler > > Sent: Thursday, September 08, 2005 11:09 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [Asterisk-Users] voice over atlantic > > > > Pay the license fee and get the GSM codec would probably be best. > > The fee is nominal and the codec is a good one... > > $0.02 > > > > W > > > > > > > > -----Original Message----- > > From: asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David > > Hajek > > Sent: Thursday, September 08, 2005 1:50 PM > > To: asterisk-users@lists.digium.com > > Subject: [Asterisk-Users] voice over atlantic > > > > Hi- > > > > I'm using IAX between two boxes, where one box is located in US and > > the second in Europe. I'm trying to achieve the best voice > quality and > > > mainly reliability between these boxes and looking for hints and > > experience of others. > > > > Facts: > > - Asterisk 1.0.7 > > - RTT varies from 130-170 ms, depends on time and actual Internet > > throughput > > > > Questions: > > - What is the sugested codec for such setup? Now I'm using > ULAW, but > > realizing it may not be the best choice. Sometimes I can > hear broken > > audio. Maybe speex is better choice? > > - Jitter buffer, yes/no? What are the suggested values. > > Currently I'm using these values: > > jitterbuffer=yes > > dropcount=10 > > maxjitterbuffer=500 > > maxexcessbuffer=300 > > minexcessbuffer=20 > > jittershrinkrate=2 > > - Trunking? Is it reliable enough? > > > > Thanks for any hints. > > > > -- > > David > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@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 > > _______________________________________________ > > --Bandwidth and Colocation sponsored by Easynews.com -- > > > > Asterisk-Users mailing list > > Asterisk-Users@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 > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@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
On Thu, Sep 08, 2005 at 04:49:49PM -0400, David Hajek wrote:> - What is the sugested codec for such setup? Now I'm using ULAW, but > realizing it may not be the best choice. Sometimes I can hear broken > audio. Maybe speex is better choice?Also consider iLBC . gsm consumes less CPU than either of those two. -- Tzafrir Cohen | tzafrir@jbr.cohens.org.il | VIM is http://tzafrir.org.il | | a Mutt's tzafrir@cohens.org.il | | best ICQ# 16849755 | | friend
Forgot the version: Asterisk 1.0.7 On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote:> Nice. Thanks. > > What Asterisk version? Can you lookup jitterbuffer settings? > > Thanks a lot.
Hi, have you tried speex? I'm going to give it a shot. I think Speex should be better then gsm. Thanks. -David> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > asterisk groups > Sent: Friday, September 09, 2005 1:53 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] voice over atlantic > > Hi David, > > I just looked at my iax.conf on one of my boxes in Argentina > and actually there are no jitterbuffer settings indicated so > I'm assuming it is using Asterisk defaults. > > We are experimenting with G.729 on these IAX trunks also and > I just realized I have no accurate means of measuring > bandwidth consumption vis-a-vis GSM/G.729. I think I'll pose > that question to the group in another message to see what > recommendations and best practices are out there. Or, do some > research. > > Best of luck. > > On Thu, 2005-09-08 at 17:49 -0400, David Hajek wrote: > > Nice. Thanks. > > > > What Asterisk version? Can you lookup jitterbuffer settings? > > > > Thanks a lot. > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@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 >