Stuart Elvish - Dallas Delta Corporation Pty Ltd
2006-May-25 22:25 UTC
[Asterisk-Users] PAP-2 Conferencing Problems
Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). I have tested the same setup with different brands of ATA's and with IP phones and there aren't any problems. I have also tested a couple of different codecs (g729 and ulaw) and the problem seems to still exist. The problem happens when the ATA is both internal and external to the VoIP server network. Does anyone have any suggestions? -------------- next part -------------- A non-text attachment was scrubbed... Name: stuart.elvish.vcf Type: text/x-vcard Size: 385 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060525/6f370b42/stuart.elvish.vcf
Stuart Elvish - Dallas Delta Corporation Pty Ltd wrote:> Just come across a problem - we have sent out heaps of PAP-2 ATA's and > just discovered that when joined in a conference they are choppy on > the up leg (so the other users in the conference will hear them with a > choppy sound) but the down leg is perfectly fine (so the end user can > hear the conference participants perfectly). >Is that into a meetme conference? If so I have noticed that you have to change the default RTP Size (on the PAP2 or Sipura) to .20 instead of .30. Don't know why that is but would be very interested in knowing if it fixes your issue. I have tried also with .60 and it is not only choppy, its totally unintelligible.> I have tested the same setup with different brands of ATA's and with > IP phones and there aren't any problems. I have also tested a couple > of different codecs (g729 and ulaw) and the problem seems to still exist. > > The problem happens when the ATA is both internal and external to the > VoIP server network. > > Does anyone have any suggestions? > > >------------------------------------------------------------------------ > >_______________________________________________ >--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 > >-- Andres Technical Support http://www.telesip.net