On Wed, Dec 30, 2009 at 8:11 AM, Thermal Wetland <thermalwetl...@gmail.com> wrote: > We have a customer on a wireless connection that has very bad jitter. They > can hear people fine, but people have a very hard time hearing them. They > are connected via a SPA-2102. > > It is a SIP client going to a SIP trunk. > > Something like this in sip.conf [general] would be in effect for all SIP > clients: > jbenable = yes > jbmaxsize = 150 > jbresyncthreshold = 1000 > jbimpl = fixed > jblog = yes > > I only want to enable the jitter buffer for the end points having the > trouble. > > Reading the docs, it seems that the jitter buffer is only used when the end > point is connected to an app like voicemail. > > -- > -Thermal > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
This is from voip-info.org - http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf It is in the [general] section # Asterisk sip jbenable = yes|no : Enables the use of a jitterbuffer on the receiving side of a SIP channel. (Added in Version 1.4) # Asterisk sip jbforce = yes|no : Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to "no". (Added in Version 1.4) It mentions the 'receiving side' which should be the incoming or upload form the clients. As I am sure you saw, it is not mentioned in the peers and clients section. Perhaps setting jbforce to no and jbimpl to adaptive. I am sure you read all that, anyone have any real world experience? Aloha, Matt _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users