Re: [asterisk-users] Re: One way choppy sound
On 1/19/07, Martin Joseph <[EMAIL PROTECTED]> wrote: On 2007-01-17 10:29:43 -0800, Yelson Vivas <[EMAIL PROTECTED]> said: > Hi Guys > I'm conecting 2 astersk servers using this arquitecture > > (Ext softphone)<==sip==>(asterisk 1)(asterisk 2) > <===alaw==>(pstn) > > If i call from the Ext to the asterisk 2 the sound is perfect, but if > i call from Ext to the pstn, i can hear perfect but they tell me that > sound really choppy, i tried using several codecs (same problem) but > i don't understand why the sound is bad in only one way. > Any sugestions to solve it more than welcome Usually sounds can be "choppy" one way due to constrained upstream bandwidth. There might be plenty of room for the audio to get to you, but that doesn't mean the reverse is at all true. Jitter buffering can help this, or using a more compact format (like GSM or g729) is also a potential helper. Good luck, hope this helps, Marty On 21 Jan 2007, at 03:46, Andrew Joakimsen wrote: I've actually found in many cases a lower bandwidth codec doesn't improve at all and however it oftentimes makes the issue worse. What other traffic do you have on the IAX trunk link ? Even if it isn't 'full' you may be hearing your IAX packets being delayed behind 'bigger' packets, or sitting in a low priority queue on a router. You might want to look into applying a QoS to the link. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --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
Re: [asterisk-users] Re: One way choppy sound
I've actually found in many cases a lower bandwidth codec doesn't improve at all and however it oftentimes makes the issue worse. On 1/19/07, Martin Joseph <[EMAIL PROTECTED]> wrote: On 2007-01-17 10:29:43 -0800, Yelson Vivas <[EMAIL PROTECTED]> said: > Hi Guys > I'm conecting 2 astersk servers using this arquitecture > > (Ext softphone)<==sip==>(asterisk 1)(asterisk 2) > <===alaw==>(pstn) > > If i call from the Ext to the asterisk 2 the sound is perfect, but if > i call from Ext to the pstn, i can hear perfect but they tell me that > sound really choppy, i tried using several codecs (same problem) but > i don't understand why the sound is bad in only one way. > Any sugestions to solve it more than welcome Usually sounds can be "choppy" one way due to constrained upstream bandwidth. There might be plenty of room for the audio to get to you, but that doesn't mean the reverse is at all true. Jitter buffering can help this, or using a more compact format (like GSM or g729) is also a potential helper. Good luck, hope this helps, Marty ___ --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 ___ --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
[asterisk-users] Re: One way choppy sound
On 2007-01-17 10:29:43 -0800, Yelson Vivas <[EMAIL PROTECTED]> said: Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)<==sip==>(asterisk 1)(asterisk 2) <===alaw==>(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call from Ext to the pstn, i can hear perfect but they tell me that sound really choppy, i tried using several codecs (same problem) but i don't understand why the sound is bad in only one way. Any sugestions to solve it more than welcome Usually sounds can be "choppy" one way due to constrained upstream bandwidth. There might be plenty of room for the audio to get to you, but that doesn't mean the reverse is at all true. Jitter buffering can help this, or using a more compact format (like GSM or g729) is also a potential helper. Good luck, hope this helps, Marty ___ --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
