Thermal Wetland
2009-Dec-30 18:11 UTC
[asterisk-users] Force Jitter Buffer for SIP to SIP calls
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091230/53455905/attachment.htm
Matt Darnell
2009-Dec-30 18:27 UTC
[asterisk-users] Force Jitter Buffer for SIP to SIP calls
On Wed, Dec 30, 2009 at 8:11 AM, Thermal Wetland <thermalwetland at 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