similar to: sip/rtp performance monitoring

Hi! I want to use Asterisk as load generator to test quality degradation with increased load (e.g. testing other SIP equipment or IP-links). Is anybody aware of such a setup with Asterisk - is it possible to get RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)? Thanks Klaus

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

Something might have slipped by me, but I got into the following situation installing Rcmdr: < install.packages("Rcmdr",depend=TRUE) > Oodles of Output, until: * Installing *source* package 'multcomp' ... ** R ** data ** inst ** preparing package for lazy loading Error in dyn.load(x, as.logical(local), as.logical(now)) : unable to load shared library

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

Dear R experts---I may have asked this in the past, but I don't think I figured out how to do this. I would like to execute traceback() automatically if my R program dies---every R programI ever invoke. I guessed that I could have wrapped my entire R code into tryCatch( ... oodles of R code , error = function(e) traceback(), finally = cat("done") } but the traceback docs tell

Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys?, This is wht i see on asterisk console?: ? -- Executing [9613070741 at direct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? --

(hello all, this is my first writing. so please bear with me if I'm wrong anywhere.) orry to break too lately, but how is the RTP payload submission is going? could we see the new payload at March IETF? I agree that it would be fairy straightforward to make an RTP payload for ogg vorbis, assuming raw packets, AFAIK. using physical bitstream is, in this case, not adequate by the reasons in

G'day. I am wanting to find out how my SIP service is performing with Asterisk, especially jitter and dropped packets. I can get an overview of that using the 'rtcp stats' function at the console, but is there any way to get those logged to a file or some other permanent record? Nothing in logger.conf seems applicable, save perhaps directing verbose messages somewhere, but it

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

Hello, Has anyone encountered that when Asterisk sends RTCP messages, it stops sending RTP packets until it gets an answer? Can that be fixed? Thanks.

Hi guys, I am working on an application that gathers and stores toll-quality/narrow-band voice data. It will allow clients to request this data and stream it to them on the fly. I'm planning on this data all being stored in the speex format (possibly encapsulated in an Ogg file header). I was wondering what method the members of this list would recommend for streaming the data to

How Magic Jack can only charge $20 per year? do they have a call limit? do they have a call duration limit or limit of minutes per day?, Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/974cf988/attachment.htm

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination

Full_Name: Anne Gilman Version: 2.8.1 OS: Mac OS X 10.4.11 Submission from: (NULL) (132.177.75.245) Dear R Team, Today I upgraded from R 2.2.2 to 2.8.1 ...then I restarted, did a bunch of work, and tried to quit. (Note that I have oodles of stuff saved in the work environment; took almost a full minute to start up.) After over 10 minutes, it was still trying to close up, and the problem

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN

Good question. I'm coming to the conclusion that using plain UDP and "home-grown" packet construction for transmitting the speex data (with timestamp/sequence counter) and implementing jitter control on the receiver end is an adequate implementation for a VoIP application. Assuming of course that I don't care about any interoperability issues with other applications etc. I was

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

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

Does anyone out there have a recommendation for tools that will monitor the quality of VoIP systems? I am looking for jitter and MOS monitoring. I have a custom Nagios plugin that is alerting me if the jitter jumps out of a 20ms but I am looking for a little more detail. I would not be against writing something in Perl for Nagios to do but I don't really know where to start on measuring jitter