search for: channelstats

...between Digium and Adtran and they can figure out why it doesn't work. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Todd R. Sent: Monday, January 19, 2015 1:45 PM To: Asterisk-Users List Subject: Re: [asterisk-users] sip show channelstats reliable? Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers...

I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss...

Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. From: EWieling at nyigc.com To: tjrlist at live.com; asterisk-users at lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I?ve seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.3...

Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter...

...issue, I just don't know how to > prove it outside Asterisk. > > > ------------------------------ > From: EWieling at nyigc.com > To: tjrlist at live.com; asterisk-users at lists.digium.com > Date: Mon, 19 Jan 2015 13:55:33 -0500 > Subject: RE: [asterisk-users] sip show channelstats reliable? > > > I?ve seen something similar with Adtran SIP gateways. When a re-invite > happens the Adtran gets all confused about call stats and marks the > pre-reinvite leg of the call as losing large numbers of packets. BTW, > IIRC reinvites happen when a codec changes o...

...w to >> prove it outside Asterisk. >> >> >> ------------------------------ >> From: EWieling at nyigc.com >> To: tjrlist at live.com; asterisk-users at lists.digium.com >> Date: Mon, 19 Jan 2015 13:55:33 -0500 >> Subject: RE: [asterisk-users] sip show channelstats reliable? >> >> >> I?ve seen something similar with Adtran SIP gateways. When a re-invite >> happens the Adtran gets all confused about call stats and marks the >> pre-reinvite leg of the call as losing large numbers of packets. BTW, >> IIRC reinvites happe...

...You can reach the person managing the list at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. sip show channelstats reliable? (Todd R.) > 2. Re: sip show channelstats reliable? (Todd R.) > 3. Re: sip show channelstats reliable? (Eric Wieling) > 4. Re: sip show channelstats reliable? (Todd R.) > 5. Re: sip show channelstats reliable? (Scott Griepentrog) > 6. Re: SEMI-OFFTOPIC openvox (ric...

Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I?ve been playing around with ?sip show channelstats? but can?t other than measuring the packet loss I don?t really know what I?m supposed to be looking for in order to say ?ah ha! that?s the problem!?. I also don?t know what it?s limits are. Will the stats in ?sip show channelstats? show a customer using a torrent client and saturating their own bro...

Hello! I'm just wondering what the RTT exactly means. Where are the exact measuring points located? > pjsip show channelstats ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT.... =====================================================================================...

...or almost a year, I decided to give plain asterisk a try, so I installed CentOS 6.4 and Asterisk 1.8. After configuring it (sip.conf, extensions.conf, even meetme.conf to try a conference room) everything seemed to work fine, except for the fact that when a call is placed, and I use the sip show channelstats command, everything shows as 0, no packets sent or received, and no packets lost, which is strange because the call is taking place normally. I'm using canreinvite=no, so I'm sure asterisk is "in the middle" of the conversation. What's even more strange, is that the command...

Hello, what could be causing the issue of poor sound quality ? Some calls, certainly not all of them, sound like if the caller is standing next to a very busy road with lots of cars passing. To be clear : the person calling is not standing next to a highway. But there seems to be a noise "on the line" of busy highroad that makes that the caller can not be understood. What can be

Google says Round Trip Time https://www.voip-info.org/asterisk-rtcp/ Doug

On Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > On 15.05.20 at 14:31 Doug Lytle wrote: > > Google says Round Trip Time > > > > https://www.voip-info.org/asterisk-rtcp/ > > That doesn't answer my question (I know the abbreviation RTT). Therefore > I'm trying again: > > I'm just wondering what the RTT *exactly*

On 17.05.20 at 01:28 Joshua C. Colp wrote: > On Sat, May 16, 2020 at 10:58 AM Michael Maier <m1278468 at mailbox.org> wrote: > >> => How are the RTT values exactly calculated? Which values are actually >> used for? >> > > The value is calculated according to the logic in the RFC[1]. Specifically > using embedded timestamps in the RTCP packets and

On 15.05.20 at 14:31 Doug Lytle wrote: > Google says Round Trip Time > > https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually

...i On Tue, Mar 31, 2015 at 1:16 PM, Patrick Beaumont < p.beaumont at hatsoffsoftware.co.uk> wrote: > Thanks for the suggestions guys. I?ll try to have a play with Voipmonitor > in the near future. > > So can I assume from the lack of discussion nobody is using the ?sip show > channelstats? stuff? > > Regards, > Patrick. > > On 31/03/2015 08:23, "Olivier" <oza.4h07 at gmail.com> wrote: > > >Some SIP hardphones (Polycom) or softphones (Counterpath) embed a > >module that metter MOS. > > > > > >Regards > > > >2...

I am not sure if its just me, but i am able to get only local channel variables containing RTCP QOS values. The Version is 1.8.14. I want to store values of bridged channel in CDR. Phone is Cisco 7941 SIP and with sip show channelstats i see all the relevant information (jitter,packet loss) i want to get. It even calculates packet loss in %. But i am not able to store it to CDR. Asterisk 1.4 seems to have had a function ast_rtp_get_quality but i cant find any information about that in sources from 1.8, only a short reference...

...; > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have any advice or links to advice on measuring call > quality? > > I?ve been playing around with ?sip show channelstats? but can?t other than > measuring the packet loss I don?t really know what I?m supposed to be > looking for in order to say ?ah ha! that?s the problem!?. I also don?t > know what it?s limits are. Will the stats in ?sip show channelstats? show > a customer using a torrent client and satu...

...; > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have any advice or links to advice on measuring call > quality? > > I?ve been playing around with ?sip show channelstats? but can?t other than > measuring the packet loss I don?t really know what I?m supposed to be > looking for in order to say ?ah ha! that?s the problem!?. I also don?t > know what it?s limits are. Will the stats in ?sip show channelstats? show > a customer using a torrent client and satu...

...my case it does not appear to be a NATing issue, since running OpenVPN from pfSense means there's no NATing occurring between the clients or between the clients and the asterisk server. Although I was unable to reproduce the problems, I did notice some packet loss and jitter in "sip show channelstats", here is a sample: Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 192.168.32.26 446613544 at 1 00:03:03 0000000094 0000004238 (97.83%) 0.0000 0000000000 0000000244 ( 0.00%) 0.0000 192.168.32.38 5b2ebdc92fd 00:03:0...