similar to: tcpdump auto stats script

We have a system where two calls are in a ConfBridge with recording. This is Asterisk 16.3.0 Channel A seems to work perfectly. Wireshark is showing the RTP to/from working fine and having no jitter/lag issues. This call hears everything from channel B. Channel B we have more issues capturing a wireshark trace because their channel can be in the system for hours. When the two calls are in the

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

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 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

Hello All, MOS and R factor are the two QoS parameters used to estimate VoIP call quality. I have found that they are calculated from other metrics like jitter, latency, packet loss,...etc. But, haven't found any formula or arithmetic rule to calculate them. Do you have an idea about their formulas or an open source that calculates them. Is it possible to interpret them from wireshark.

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.

Hi! I am looking to calculate the end-to-end delay between two soft phone/hard phone. I have asterisk server and configured ntp server on the same machine and synchronized it with ntp pool. I have seen that Wireshark can be used to check the jitter. But I am not sure how can i calculate the end to end. May be this is not related to the mailing list topic but please help me if anyone has some

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

Guys, I'm having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from people. Soon I figured some network problem, I did a tcpdump and analyzed by wireshark ...the strange thing is this ... all packets that

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 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

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

I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

Hello list, what can I do to minimalize the jitter in SIP-calls at server level ? If at local network level, there is a VoIP-router and their is a physical network dedicated to IP-phones, but there is still jitter. When using a Hosted Asterisk server, which settings on the Asterisk-server can minimalize the jitter between the VoIP-router and the Asterisk-server on the public internet ?? Kind

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

Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. It looks like there is no RTCP/RTP sent to the client from Asterisk while on hold (music on hold playing to caller) thus client disconnects the call. During this time, I get the following messages in the CLI: NOTICE[24194] rtp.c: Unknown RTP codec 126

When on a call, you can press the middle round button and bring up some RTP statistics. Can anyone confirm my theory that the AvgJtr and MaxJtr are between this phone and the far end? Or is this jitter reading only between this phone and asterisk? I'm guessing its the foremost, because when I make a local call to PRI, the jitter is low/0 since the call would terminate at asterisk. But

Hi, Does anybody know any software that give me Latencty, Jitter and Lost packets to analyze my Call quality ??? Luis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070831/47350d13/attachment.htm

I'm currently running asterisk to provide VoIP services to clients of the ISP I work for. I would like to be able to tell if I am loosing packets and/or are having other issues with any of the voice streams, so I can address them proactively. I'm not particularly interested in spending oodles of money buying one of the commercial analysis tools. Is there some open source tool (or

Hello, Our Asterisk runs with multiple remote sites (12 over an MPLS network), everything works fine except for the last site we have juste installed. When VOIP flows comes/goes from/to this site, there are sound quality issues, persistent, 100% reproducible, on every call. This is not a bandwidth or latency or jitter problem, everything is fine on the network. Our MPLS provider does all check