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)<====iax2 trunk====>(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
