Thanks for the tips. The Grandstream doesn't have a G711 or uLaw option for codecs. It has PCMU, PCMA and iLBC. Are any of these related to G711 ?
Grandstreams have echo cancellation and it appears to be working after a few seconds of conversation. What is VAD ? On Thu, 2005-01-27 at 20:22 -0500, asterisk lists wrote: > Try using g711 (ulaw) and make sure to turn Silence Suppression OFF as > asterisk needs the full audio stream for assembling the audio streams. > Once you get the call quality good using g711 (ulaw), then you can > play around with the other codecs (g729, etc). Also, try a 20ms frame > size. > > Unfortunately, echo is usally introduced at the central office due to > an impedence imbalance. Some SIP phones have echo cancellation > options built-in to compensate (not sure the Grandstream has that > feature). > > You may also see if you have VAD enabled in the phone. If you do, turn it > OFF. > > Hope that helps! > > - Pedro > VoIP by TRACI.net > > > On Thu, 27 Jan 2005 08:53:02 -0700, Kim Lux <[EMAIL PROTECTED]> wrote: > > I'm testing a bunch of stuff before we implement our system. > > > > I've got a SIP account and Grandstream phones. We haven't started using > > asterisk yet. Generally we've got good voice quality from all the > > offices except: > > > > a) We get a lot of echo in the first 10 seconds or so of the call, only > > on the VOIP calling end. The callee says the speech sounds normal. To > > the caller, the first Hello is almost intelligible with echo. > > > > b) The first part of an abrupt statement from one party gets "clipped". > > In conversation, when talking switches from one party to the other, a > > tiny bit of speach gets clipped. > > > > c) If both parties talk at once there is a bit of dropout. > > > > We'd like to improve the voice quality in these respects. Otherwise the > > voice quality is excellent. I've been told it is better than the > > traditional system several times. > > > > Questions: > > > > a) Are certain codecs better than others at quickly getting the echo > > cancellation setup ? Is there a way to get the echo out of the call > > immediately ? (Is there a document explaining the features and pitfalls > > of all the codecs somewhere ?) > > > > b) Is there a way to eliminate the speech clipping when speakers change > > or both talk at once ? I've read about asterisk injecting noise and/or > > sending packets in the absence of speech. Would that help ? Is this > > what the Grandstream "Silence Suppression" is about ? > > > > c) How does one know where to set the following: > > > > iLBC frame size: 20ms 30ms > > iLBC payload type: (between 96 and 127, default is 98) > > Silence Suppression: No Yes > > Voice Frames per TX: (up to 10/20/32/64 for G711/G726/G723/other codecs > > respectively) > > Layer 3 QoS: (Diff-Serv or Precedence value) > > Layer 2 QoS: 802.1Q/VLAN Tag 802.1p priority value (0-7) > > > > d) One place we've really got a problem is when we use a Grandstream in > > a big echoy (sp!) room. We seem to get echo from the room into the call > > which seems to fool the echo cancellation. Any ideas on how to get > > around this problem ? > > > > d) How is asterisk going to change our sound quality when it is added > > between the phones and the SIP provider ? Does it have features that > > will help with the echo and clipping and if so, how much improvement > > should we expect ? > > > > Thanks. > > > > -- > > Kim Lux, Diesel Research Inc. > > > > _______________________________________________ > > 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 > > > _______________________________________________ > 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 > -- Kim Lux, Diesel Research Inc. _______________________________________________ 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