[asterisk-users] Understanding CHANNEL function values
Hello all, I need some help understand the values of the CHANNEL function, e.g. txploss // local packets loss rxploss // remote packets loss txjitter // local jitter rxjitter // remote jitter My main problem in understand is that a CHANNEL has two nodes (sender and receiver), while a typical setup includes at least 3 nodes: SIP phone - Asterisk - SIP Provider ( - each is a node) 1) So e.g. txploss, is it - what is lost between SIP phone and Asterisk - what is lost between Asterisk and SIP Provider - or probably both? I guess the SIP Provider sends back a info about missed packets, then it wouldn't be relevant where they got lost, but just that they were lost somewhere between the SIP phone and the SIP Provider? 2) Also, how can I monitor only the connections SIP phone - Asterisk and Asterisk - SIP Provider each on their own? 3) txploss are the lost packets in the direction from SIP phone to SIP provider, right? I am aware of tx and rx and what it normally means, but in this case it would also fit the other way round ;-) 4) I always have txjitter but never rxjitter, does this make sense? Shouldn't txjitter be less of a problem? Also, how exactly is the txjitter / jitter in general defined in Asterisk? Thanks very much and best regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding CHANNEL function values
On 25-08-12 14:31, Stefan at WPF wrote: Hello all, I need some help understand the values of the CHANNEL function, e.g. txploss // local packets loss rxploss // remote packets loss txjitter // local jitter rxjitter // remote jitter My main problem in understand is that a CHANNEL has two nodes (sender and receiver), while a typical setup includes at least 3 nodes: SIP phone - Asterisk - SIP Provider ( - each is a node) 1) So e.g. txploss, is it - what is lost between SIP phone and Asterisk - what is lost between Asterisk and SIP Provider - or probably both? I would assume that those statistics apply to a leg and not an end-to-end connection. So in your example I would assume that a txploss value is determined for the leg between the SIP phone and the Asterisk server and another txploss value is determined for the leg between the Asterisk server and the upstream SIP provider. Interesting stuff. If you figure it all out, please update this thread (and possibly the wiki). Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding CHANNEL function values
Hello Patrick, there are the channel and dstchannel properties of the CHANNEL function, indicating that the channel is some kind of virtual connection between the sip phone and the sip provider, so it seems not to be for legs or at least the output values are already a combination of both legs. (I also would find it more intuitive to have 2 legs and values for those instead of one single value for 2 legs, but lets wait to find out, how to interpret those things). 2012/8/25 Patrick Lists asterisk-l...@puzzled.xs4all.nl On 25-08-12 14:31, Stefan at WPF wrote: Hello all, I need some help understand the values of the CHANNEL function, e.g. txploss // local packets loss rxploss // remote packets loss txjitter // local jitter rxjitter // remote jitter My main problem in understand is that a CHANNEL has two nodes (sender and receiver), while a typical setup includes at least 3 nodes: SIP phone - Asterisk - SIP Provider ( - each is a node) 1) So e.g. txploss, is it - what is lost between SIP phone and Asterisk - what is lost between Asterisk and SIP Provider - or probably both? I would assume that those statistics apply to a leg and not an end-to-end connection. So in your example I would assume that a txploss value is determined for the leg between the SIP phone and the Asterisk server and another txploss value is determined for the leg between the Asterisk server and the upstream SIP provider. Interesting stuff. If you figure it all out, please update this thread (and possibly the wiki). Regards, Patrick -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
