Non-Immersive Endpoints Call Statistics Reports - jitter, packet loss

Hi,
We are trying to have reports from calls of Non-Immersive Endpoints. We need to have info about jitter and packet loss.
I know it is saved in the logs in file called call_history.txt and available via Web interface under "Diagnostics - Call History".
Issue is how to collect/download such data from Non-Immersive Endpoint automatically. I am not talking about download logs, extract the files, go to specific folder and open specific file.
We have such reports for Immersive Endpoints because Call Statistics are included in MIB.
Any idea/experience to collect such data from Non-Immersive Systems?
Many thanks,
Josef

@Raju_raju
Many thanks for this - its what I expected but its still disappointing to hear. I think Microsoft are missing a trick here, even WebRTC (in Chrome at least) and most other conferencing applications have call statistics built in.
The lack of true real time monitoring in Lync out of the box worries me somewhat. I know that there might be third party options, including Microsoft's own SCOM (which of course, not everybody uses), but I would like to see some real time monitoring of this
real time system as part of its native functionality. True stats available at the client would still be a great help - even if it meant a ALT+Click type procedure.
I need to dig deeper into the normal Lync monitoring options but I'm used to environments where we can analyse each "video hop" as often, there are specific network segments that cause glitches. For instance, in a call where media is routed via
an edge, I would liek to see the stats on the leg from the client to the edge then from the edge to the FEP or Director etc.
Thanks for the heads up on the Snooper documentation. I have read this page previously but will re-visit, yet this is somewhat over kill for some basic real time statistical output.
@Eric - Great table. I must say I have never come across this and it certainly has useful stuff. I can almost understand why Microsoft shows the simplistic feedback by default, but for an advanced user or engineer having to look up the dumbed down feedback
against such a table is simply a waste of time. The actually stats (including both percentage and raw packet loss data) is eminently more useful and immediate.
Microsoft - are you listening?
Chris

Similar Messages

TMS reporting - Network Packet Loss

Hi everyone,
I am trying to see packet loss reports in TMS but there is no info there.
My network is : VCS 7.0.2 with movi, c20 and ex60 registered (SIP) to it and a TMS 13.1.2.
The vcs is a starter pack and the TMS is the demo version. I first registered the endpoint to the vcs and then added the TMS to the network. I the added the endpoints and vcs to TMS. In TMS I see the endpoints and info about them, but unfortunatelly no info in the packet loss report. I made a call between movi and c20 with packet loss but no info was shown in the report.
I am very new to TMS and I wonder if there is something I should do in order to make the endpoints send packet loss info to TMS.
I sshed to the endpoints and I can see the info about packet loss and jitter through the api.
The network described above is my test network on which I can do prety much everything on. I also have the same problem (this time with EX60 endpoints, VCS Control and TMS) on a customer network and on that network I really need to get this working.
Any help would be much appreciated. Form what I read the endpoints should sent this info through HTTP to TMS. Am I right?
Thank you for your help.

Hi Magnus,
Thank you for your reply. I checked the logs and I get some info over there. I have attached the log.
Do you know what doc should I check to see if Ex60 is supported?
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/8/2012 3:58:51 AM
Disconnected
Call id: 3 Port id: 1;
Remote Number: sip:[email protected] Remote Name: 1853;
Direction: Outgoing CallProtocol: SIP;
Bandwidth: 6000 kbps Encryption Mode: Aes128;
Duration: 00:13:33; Cause Code: Normal call clearing
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/8/2012 3:45:19 AM
Connected
Call id: 3 Port id: 1;
Remote Number: sip:[email protected] Remote Name: ;
Direction: Outgoing CallProtocol: SIP;
Bandwidth: 6000 kbps Encryption Mode: Off;
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/7/2012 6:43:15 AM
Disconnected
Call id: 2 Port id: 1;
Remote Number: sip:[email protected] Remote Name: 1853;
Direction: Outgoing CallProtocol: SIP;
Bandwidth: 6000 kbps Encryption Mode: Aes128;
Duration: 00:14:57; Cause Code: CauseCode_Unknown
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/7/2012 6:28:19 AM
Connected
Call id: 2 Port id: 1;
Remote Number: sip:[email protected] Remote Name: ;
Direction: Outgoing CallProtocol: SIP;
Bandwidth: 6000 kbps Encryption Mode: Off;
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/7/2012 6:28:07 AM
Got Response
Got Network Response - TMS will now monitor   the system.;
System: [email protected] -   10.1.x.y ;
System Status: No response;
System Type: TANDBERG C-series   Endpoint;
Software: s52000 TC5.1.0.280662;
Hardware: F1AN19D00469 - ;
System Contact;
No Contact ();
Mobile: ;
Office: ;
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/7/2012 6:28:07 AM
Got Response
Got Network Response - TMS will now monitor   the system.;
System: [email protected] -   10.1.x.y ;
System Status: Idle;
System Type: TANDBERG C-series   Endpoint;
Software: s52000 TC5.1.0.280662;
Hardware: F1AN19D00469 - ;
System Contact;
No Contact ();
Mobile: ;
Office: ;
10.1.x.y
TANDBERG Codec C20
Other type
Endpoint
5/7/2012 6:28:06 AM
Got Response
Got Network Response - TMS will now monitor   the system.;
System: [email protected] -   10.1.x.y ;
System Status: No response;
System Type: TANDBERG C-series   Endpoint;
Software: s52000 TC5.1.0.280662;
Hardware: F1AN19D00469 - ;
System Contact;
No Contact ();
Mobile: ;
Office: ;

IPSLA-ICMP Jitter-Packet loss count

Hi,
I wanted to monitor the packet loss in my link using IP SLA. So i configured ipsla jitter and started monitoring it. When i analyze the output i found that ip sla statictics shows heavy drop in my link, but when i ping from the router no drop is observed. Why there is such difference in the out put. I read that normal ping and icmp jitter uses different icmp types messages but now i am confused whether my link proper or not. Which data i should belive.
Here is my config and statistics:
ip sla 2166
icmp-jitter 10.70.194.137 num-packets 100
frequency 300
ip sla schedule 2166 life forever start-time now
Type of operation:              icmpJitter
        Latest RTT: 7 milliseconds
Latest operation start time: 13:56:49.468 IST Sat Dec 24 2011
Latest operation return code: OK
RTT Values:
        Number Of RTT: 92               RTT Min/Avg/Max: 3/7/158
Latency one-way time:
        Number of Latency one-way Samples: 0
        Source to Destination Latency one way Min/Avg/Max: 0/0/0
        Destination to Source Latency one way Min/Avg/Max: 0/0/0
Jitter Time:
        Number of Jitter Samples: 89
        Source to Destination Jitter Min/Avg/Max: 1/4/20
        Destination to Source Jitter Min/Avg/Max: 1/1/1
Packet Late Arrival: 0
Out Of Sequence: 0
        Source to Destination: 0        Destination to Source 0
        In both Directions: 0
Packet Skipped: 0
Packet Loss: 8
        Loss Period Length Min/Max: 1/7
Number of successes: 1
Number of failures: 0
Operation time to live: Forever

Hello mate !
For how long you left the IP sla running ? 24 hours ?
I would disable session time out in your router, set the IP Sla running for 60 minutes for example and in the meantime i would leave a ping running too with high repetition in the background.
I saw you only had 8 pkt loss....   we need two "tools" running for a good comparison, tried what i proposed and let me know.
cheers

100% packet loss

Hi there,
I get 100% packet loss whilst I am playing online on the Xbox, it only occurs on FIFA 13, I have contacted EA and they have said the packet loss occurs within the BT network, I was wondering if anyone else has the same problem and whether you could help me? Here is my trace route to the EA servers [Easo.ea.com];
Host Name IP Address Hop Ping Time Ping Avg % Loss Pkts r/s Ping best/worst
BTHomeHub.home 192.168.1.254 1 1ms 2ms 0% 37 / 37 1ms / 41ms
* Unknown Host * 217.32.147.102 2 6ms 7ms 0% 37 / 37 6ms / 12ms
* Unknown Host * 217.32.147.142 3 8ms 8ms 0% 37 / 37 6ms / 32ms
* Unknown Host * 212.140.206.82 4 11ms 11ms 0% 37 / 37 10ms / 15ms
* Unknown Host * 217.41.169.221 5 10ms 10ms 0% 37 / 37 10ms / 16ms
* Unknown Host * 217.41.169.109 6 11ms 11ms 0% 37 / 37 10ms / 22ms
* Unknown Host * 109.159.251.211 7 12ms 100% 0 / 37
core1-te-0-0-0-17.ilford.ukcore.b109.159.251.189 8 20ms 100% 0 / 37
core1-pos9-1.telehouse.ukcore.bt.194.74.65.114 9 17ms 100% 0 / 37
166-49-211-153.eu.bt.net 166.49.211.153 10 18ms 26ms 0% 37 / 37 18ms / 191ms
t2a1-ge4-0-0.uk-lon1.eu.bt.net 166.49.135.17 11 15ms 17ms 0% 37 / 37 15ms / 28ms
te0-7-0-3.ccr21.lon01.atlas.cogen130.117.14.49 12 20ms 19ms 0% 37 / 37 18ms / 21ms
te0-3-0-5.mpd21.lon13.atlas.cogen130.117.0.58 13 20ms 100% 0 / 37
te0-0-0-4.ccr21.bos01.atlas.cogen154.54.5.161 14 94ms 100% 0 / 37
* Unknown Host * 154.54.5.98 15 99ms 100% 0 / 37
te0-1-0-5.mpd21.iad02.atlas.cogen154.54.5.46 16 93ms 100% 0 / 37
* Unknown Host * 154.54.88.173 17 100ms 124ms 0% 37 / 37 97ms / 459ms
* Unknown Host * 154.54.88.173 18 99ms 114ms 0% 37 / 37 97ms / 200ms
fert07-eqx-iad.ea.com 159.153.224.226 19 94ms 100% 0 / 36
fert07-eqx-iad.ea.com 159.153.224.226 20 97ms 100% 0 / 36

Can you confirm what test you ran that gave you these statistics?
100% packet loss would mean that there is nothing coming through the pipe for that period of time. You are essentially disconnected at 100% loss, speed is the least of your concerns. That is also going to depend on what you are testing your packet loss test against. Perhaps VZW is blocking this particular test from completing.
A more standard test is to get a browser, connect to the Jetpack and navigate to SpeedTest.net. Run a few tests here and take the average speeds that are recorded to paint a better picture.
Xbox also has well known limitations with the VZW 4G LTE network. Unless something new has happened all Xbox users on 4G LTE are not able to host multiplayer games due to the NAT firewall that runs that network.

Packet marking verification of outgoing packets for Immersive Endpoints TX, CTS Series

Hi,
Is there any way how to verify outgoing packet marking from TX and CTS Immersive systems on the system itself please?
Can I find it somewhere in the logs? In which one please?
I can see packet marking for incoming packets in Call Statistics but not for outgoing packets.
CUCM config is EF for voice and AF41 for Video (verified at Telephony Settings) but I can see both audio and video incoming packets on other site as AF41.
Any answer really appreciated.
Regards,
Josef

As Chris said, use something with port mirroring to be sure.
There is also a capture cpapability on the CTS / TX systems, please see the cli guide
utils network capture
utils network capture [dest ip-address-or-hostname]
[host [ ip | arp | rarp | tcp | udp | all]://ip-address-or-hostname] [page][numeric][hex][file][count number][size number][port number] [src ip-address-or-hostname]
Syntax Description
count number dest
ip-address-or-hostname file
hex
host
protocol://ip-address-o r-hostname
numeric page
port number size number
src
ip-address-or-hostname
Optional. Number count of packets to capture. Maximum count is 1000 for file, 10000 for screen.
Optional. Source address of the packet as a dotted IP or hostname.
Optional. Sends capture output to a file (netCapture).
Optional. Shows output in hex and ascii.
Optional. The all keyword specifies all protocols. ip-address-or-hostname is the host address of the packet and can be a dotted IP or hostname. The :// is required between the protocol and IP address or host name.
Optional. Show hosts as dotted IP addresses.
Optional. Pauses output.
Optional. Port number of the packet (either src or dest).
Optional. Number of bytes of the packet to capture. Maximum size is 1024. Optional. Source address of the packet as a dotted IP or hostname.

Packet loss higher than 100% in SQL Reporting

One of our customers is having network problems and packet loss in their network - according to Lync. The network guys claim these values are unrealistic and there've been no dropped packets on their WAN network.
On the other side, I'm looking at a Call Details Report from Lync SQL Reporting and see this in the gateway leg information:
Audio Stream (Caller -> Callee)
Codec:
PCMU
Sample rate:
8000
Packet utilization:
285
Avg. packet loss rate:
51.20%
Max. packet loss rate:
79.52%
Avg. jitter:
0 ms
Max. jitter:
0 ms
Burst duration:
5980 ms
Burst gap duration:
1890 ms
Burst density:
100.00%
Burst gap density:
0.00%
Avg. concealed samples ratio:
60.00%
Avg. stretched samples ratio:
0.00%
Avg. compressed samples ratio:
0.00%
Avg. network MOS:
1.50
Min. network MOS:
1.50
Avg. network MOS degradation:
2.21
Max. network MOS degradation:
2.21
NMOS degradation (jitter):
0.00%
NMOS degradation (packet loss):
100.00%
Audio Stream (Callee -> Caller)
Codec:
PCMU
Sample rate:
8000
Audio FEC:
False
Bandwidth estimates:
Packet utilization:
146
Avg. packet loss rate:
900.00%
Max. packet loss rate:
900.00%
Avg. jitter:
1540 ms
Max. jitter:
4406 ms
Avg. round trip:
178 ms
Max. round trip:
179 ms
From this I feel there are things to troubleshoot. :-) but how come that Lync reports a 900% packet loss?
I thought that a 100% packet loss was everything... can someone tell me what exactly are these 100+ percentages?
Thanks in advance.

latest patch level :) are you not seeing these values in your Server Performance reports?
The other thing that bugs me is when you have 1 call for 1 second with "900% loss", that would average down to get red values in the summary columns for the other 100 calls as well, as the calls are not counted for seconds, but just piece by piece.
So 8 perfect calls to this one above would average "100% packet loss per call" in the summary, which is just unusable...

VoIP Phones - Testing Latency, Jitter, and Packet Loss

I am having big problems with my VoIP phone connection and I'll try to lay it out clearly here.
The main telephone system resides at Location A (static IP address - see below - xxx.xxx.206.19), which has a network connection of 50MB down/20MB up (i.e., very fast). The VoIP phone configured for that system resides at Location B, which has a network connection of 10MB down/1MB up (i.e., also fast, or at least fast enough "on paper" for a quality VoIP connection). The LAN at Location A uses an Airport Extreme router, which does not have QOS or EF capability. The LAN at Location B uses a D-Link DIR-655 router which does have QOS that is configured properly to direct all traffic to the VoIP phone's IP address.
The VoIP phone at Location B is having intermittent call quality problems with skipping of words, hollowing out noises, jittery conversations, etc. All the inquiries I've made to the ISPs and phone system manufacturer (ESI) suggest that my base Internet speeds are not the problem.
I'm told, instead, that the problem might be latency, jitter, or packet loss between Location A and Location B. This leads to several questions:
(1)     Is there any Mac software that can test latency, jitter, and packet loss? I've looked at Network Utility and it seems to only measure a few things.
(2)     Does anyone see anything in the following Traceroute and Ping results (done twice from Location B to Location A) that looks problematic to VoIP quality?:
Traceroute:
First run: Traceroute has started…
traceroute to xxx.xxx.206.19 (xxx.xxx.206.19), 64 hops max, 72 byte packets
1 alfirving (192.168.0.1) 0.569 ms 0.363 ms 0.302 ms
2 10.72.28.1 (10.72.28.1) 27.567 ms 18.161 ms 22.288 ms
3 70.125.216.150 (70.125.216.150) 9.841 ms 10.346 ms 9.497 ms
4 24.164.209.116 (24.164.209.116) 11.042 ms 8.298 ms 9.433 ms
5 70.125.216.108 (70.125.216.108) 21.068 ms 20.657 ms 12.045 ms
6 te0-8-0-2.dllatxl3-cr01.texas.rr.com (72.179.205.48) 11.154 ms 11.540 ms 24.495 ms
7 107.14.17.136 (107.14.17.136) 11.994 ms 14.217 ms 15.816 ms
8 ae-3-0.pr0.dfw10.tbone.rr.com (66.109.6.209) 14.566 ms 32.670 ms 15.947 ms
9 ix-0-3-2-0.tcore2.dt8-dallas.as6453.net (209.58.47.105) 11.647 ms 12.260 ms 12.386 ms
10 if-2-2.tcore1.dt8-dallas.as6453.net (66.110.56.5) 10.023 ms 12.285 ms 12.338 ms
11 209.58.47.74 (209.58.47.74) 17.641 ms 16.741 ms 16.372 ms
12 0.ae2.xl3.dfw7.alter.net (152.63.97.57) 11.584 ms 12.315 ms 12.890 ms
13 0.so-6-1-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.90) 13.812 ms
    0.ge-3-0-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.17) 18.831 ms
    130.81.23.164 (130.81.23.164) 14.189 ms
14 p14-0-0.dllstx-lcr-05.verizon-gni.net (130.81.27.40) 14.561 ms 13.621 ms 15.544 ms
15 * * *
16 static-xxx.xxx.206.19.dllstx.fios.verizon.net (xxx.xxx.206.19) 23.125 ms 24.136 ms 22.411 ms
Second run: Traceroute has started…
traceroute to xxx.xxx.206.19 (xxx.xxx.206.19), 64 hops max, 72 byte packets
1 alfirving (192.168.0.1) 0.603 ms 0.420 ms 0.324 ms
2 10.72.28.1 (10.72.28.1) 40.494 ms 26.625 ms 14.152 ms
3 70.125.216.150 (70.125.216.150) 9.431 ms 9.660 ms 9.018 ms
4 24.164.209.116 (24.164.209.116) 16.293 ms 12.339 ms 19.252 ms
5 70.125.216.108 (70.125.216.108) 15.801 ms 11.438 ms 12.068 ms
6 te0-8-0-2.dllatxl3-cr01.texas.rr.com (72.179.205.48) 23.221 ms 30.459 ms 17.519 ms
7 107.14.17.136 (107.14.17.136) 14.611 ms 15.696 ms 15.775 ms
8 ae-3-0.pr0.dfw10.tbone.rr.com (66.109.6.209) 17.643 ms 14.812 ms 16.294 ms
9 ix-0-3-2-0.tcore2.dt8-dallas.as6453.net (209.58.47.105) 11.169 ms 12.374 ms 9.849 ms
10 if-2-2.tcore1.dt8-dallas.as6453.net (66.110.56.5) 16.453 ms 12.168 ms 12.384 ms
11 209.58.47.74 (209.58.47.74) 18.015 ms 14.867 ms 16.432 ms
12 0.ae2.xl3.dfw7.alter.net (152.63.97.57) 11.471 ms 11.993 ms 12.395 ms
13 0.ge-6-3-0.dfw01-bb-rtr1.verizon-gni.net (152.63.96.42) 14.077 ms 29.153 ms
    0.ge-3-0-0.dfw01-bb-rtr1.verizon-gni.net (152.63.1.17) 17.962 ms
14 p14-0-0.dllstx-lcr-05.verizon-gni.net (130.81.27.40) 14.629 ms 12.297 ms 12.839 ms
15 * * *
16 static-xxx.xxx.206.19.dllstx.fios.verizon.net (xxx.xxx.206.19) 24.976 ms 22.170 ms 22.376 ms
Ping:
First Run: Ping has started…
PING xxx.xxx.206.19 (xxx.xxx.206.19): 56 data bytes
64 bytes from xxx.xxx.206.19: icmp_seq=0 ttl=242 time=22.814 ms
64 bytes from xxx.xxx.206.19: icmp_seq=1 ttl=242 time=24.621 ms
64 bytes from xxx.xxx.206.19: icmp_seq=2 ttl=242 time=24.711 ms
64 bytes from xxx.xxx.206.19: icmp_seq=3 ttl=242 time=24.109 ms
64 bytes from xxx.xxx.206.19: icmp_seq=4 ttl=242 time=23.336 ms
64 bytes from xxx.xxx.206.19: icmp_seq=5 ttl=242 time=25.644 ms
64 bytes from xxx.xxx.206.19: icmp_seq=6 ttl=242 time=27.755 ms
64 bytes from xxx.xxx.206.19: icmp_seq=7 ttl=242 time=25.135 ms
64 bytes from xxx.xxx.206.19: icmp_seq=8 ttl=242 time=22.443 ms
64 bytes from xxx.xxx.206.19: icmp_seq=9 ttl=242 time=24.635 ms
--- xxx.xxx.206.19 ping statistics ---
10 packets transmitted, 10 packets received, 0.0% packet loss
round-trip min/avg/max/stddev = 22.443/24.520/27.755/1.448 ms
Second Run: Ping has started…
PING xxx.xxx.206.19 (xxx.xxx.206.19): 56 data bytes
64 bytes from xxx.xxx.206.19: icmp_seq=0 ttl=242 time=27.183 ms
64 bytes from xxx.xxx.206.19: icmp_seq=1 ttl=242 time=24.629 ms
64 bytes from xxx.xxx.206.19: icmp_seq=2 ttl=242 time=22.511 ms
64 bytes from xxx.xxx.206.19: icmp_seq=3 ttl=242 time=39.620 ms
64 bytes from xxx.xxx.206.19: icmp_seq=4 ttl=242 time=26.722 ms
64 bytes from xxx.xxx.206.19: icmp_seq=5 ttl=242 time=23.183 ms
64 bytes from xxx.xxx.206.19: icmp_seq=6 ttl=242 time=25.171 ms
64 bytes from xxx.xxx.206.19: icmp_seq=7 ttl=242 time=24.412 ms
64 bytes from xxx.xxx.206.19: icmp_seq=8 ttl=242 time=23.837 ms
64 bytes from xxx.xxx.206.19: icmp_seq=9 ttl=242 time=23.785 ms
--- xxx.xxx.206.19 ping statistics ---
10 packets transmitted, 10 packets received, 0.0% packet loss
round-trip min/avg/max/stddev = 22.511/26.105/39.620/4.713 ms
(3) Any other ideas on what my call quality problem might be, or how I can tweak it? For example, would putting a DIR-655 router at Location A and enabling QOS really make a difference?
Thanks to everyone, and I hope this is not too long or difficult to understand.

Hey thanks for your reply Yeah im only getting 1 ro sometimes 2 bars reception so hopefully the antenna will beef things up but I think it is what it is perhaps.

CUCM Security - Are Jitter, Latency and Packet Loss Stats in Clear Text ?

I've reviewed the Cisco Unified Communications Manager Security Guide and see no suggestion that implementing encryption prevents MoS scoring. In other words, when you implement encryption, do you still have access to jitter, latency and packet loss information in clear text?
TIA,
Amir

Well.. the phones measure the statistics.
The phones then report the stats to CUCM for inclusion in the CMRs.
... as I recall this is done via SCCP.
So yes, it's clear text, unless your signalling is encrypted generally. On most clusters it's not.
Aaron

WIFI Packet Loss/Jitter MacBook Air 11 & LION OSX 10.7.1

I started using computers more than 30 years ago when I was 5, I had my first IBM PC at the age of 10 and have never had any interest in paying over the odds for an Apple mac mainly because I saw the Mac as a kind of "Can't open nothing", one mouse buttoned retard of the computer world.
That is, until now.
I set up an online business 3 years ago and rented a dedicated server and set the whole business up in a cloud, so to speak. Having done that, all I needed a laptop for was a remote desktop connection and to run a SIP phone (Internet phone).
My PC based laptops had almost nothing installed on them, and I wasn't using software on the laptop itself, I was using remote desktop, so - why not try a MAC? The new Macbook Air 11 is small, light and made of metal and glass so should be robust enough to travel with me.
I have to say, this was the worst move I've ever made.
I opened the new shiny macbook and the first thing I noticed was that the internet seemed hit and miss. Moving around the room I managed to find a spot whereby pages woud load quickly. Strange, my £200 acer laptop was sh.t fast everywhere in the house. No matter, I packed it back away and carried on working on the Windows machine.
I've come to Newquay this week, and i've started to try and use the Macbook again in a hotel. The wireless signal in the room is low, and speedtest shows about 1 meg down and 3/4 meg up. That may sound bad to you, but remote desktop uses about 5k/sec (modem dial up speed) and the softphone, well, my Asterisk VOIP setup is confugured to use the GSM codec so that's 8k/sec each way + overheads.
This whole setup was deliberately designed to be "thin" so I can travel with ease and work on bad connections like USB internet sticks.
Anyhow, the Macbook was unable to hold a stable connection to the remote desktop or SIP phone, even though the speed test showed a whopping 1 meg up and down. What you may not be aware of is that there is more to a connection than the speed, there is the quality as well. How many packets are lost / how much "jitter" is on the line.
Anyhow, we're not living in the 3rd world, I ran a PINGTEST and it showed a small amount of jitter but told me the line was class B, online games may suffer but voip should be fine.
I unpacked the Acer, placed it in exactly the same spot as the Apple had sat in and it worked beautifully with 1 bar of wireless signal, all day long. Phone calls were clear.
So you know now what I'm thinking. I'm sitting here with my £200 acer because I can't use the £1500 macbook air 11. I paid nearly sixteen hundred pounds for this piece of .... and it doesn't ...... work. Time to contact Apple support.
2nd Mistake!
Representitive 1: - Told me that I can't compare the Macbook Air to the Acer, the Acer has Google Chrome and everybody knows Google Chrome is the fastest browser. I was told there was nothing more he could do, its probably a bad line at the hotel. When I explained the Acer works fine for voip I was told well, maybe it is getting a better signal. I explained the Acer has the cheapest possible parts inside it and paid 1500 for this macbook, expecting it to have quality parts inside and was told I'd paid for the size, because its so small but its not considered "powerful". Apple do you train your staff? Clearly not.
Thank god I wasn't paying to talk to this moron.
Representitive 2: - Had no idea what packet loss or Jitter was, got me to do a speedtest and said that looks fine. Then he got me to remove the WIFI adapter and re add it in the network settings.
Guess what, nothing changed, its still the same hardware and software.
Representitive 3: - Still not really understanding "quality" issues with the networking interface, I was asked to install the latest Java client. I did it, only because I wanted to comply with Apples wishes so they'd help me, but they weren't helping and Java has nothing to do with the network adapter, so that was useless advice too.
Apple seem to have no idea there is a problem, even though Google has pages and pages of people saying the same as me, and their own discussion forums have thousands of people complaining https://discussions.apple.com/thread/2664670?start=0&tstart=0
Finally, late yesterday whilst speaking to d.ck head number 3 at Apple support, we found a forum post talking about a fix, 10.7.1 update. I told d. head number 3 about the update and he suggested I applied it. So I did and everything looked great, for a whole evening.
This morning, I switched on again and the same thing, slow remote desktop, choppy unusable phone. Remember the phone needs less than 20k for a conversation, thats 0.2 meg. Speedtest again showing a whole meg both ways.
I called apple support again, this time being a little forceful, and I've asked for this to be escalated, but the bottom line is - they have no fix, they don't aknowledge this as a problem and I was told LION is new, so maybe it's got a bug..
I told the guy on the phone this is a network driver issue, the intermittency of the problem shows that and the Apple's lack of settings for the network adapter means the unit is autonegotiating with the router and choosing speed and duplex settings on its own. Sometimes it does that correctly, other times not and the connection although fast has a lot of noise / packet loss / corruption.
I've found a workaround, you put the unit to sleep and wake it up again and it runs fast until the next shutdown. Not really acceptable seeing as I was paying for "the cream of the crop".
I will definately not be recommending Apple products, and i'll certainly not be replacing the Windows laptops in my business with Apple's toytown system- i'd go out of business if I had to rely on this.
All there is left now, is to look at Boot camp and see if I can wipe this waste of space linux hack from the unit and install Windows 7.

I started using computers more than 30 years ago when I was 5, I had my first IBM PC at the age of 10 and have never had any interest in paying over the odds for an Apple mac mainly because I saw the Mac as a kind of "Can't open nothing", one mouse buttoned retard of the computer world.
That is, until now.
I set up an online business 3 years ago and rented a dedicated server and set the whole business up in a cloud, so to speak. Having done that, all I needed a laptop for was a remote desktop connection and to run a SIP phone (Internet phone).
My PC based laptops had almost nothing installed on them, and I wasn't using software on the laptop itself, I was using remote desktop, so - why not try a MAC? The new Macbook Air 11 is small, light and made of metal and glass so should be robust enough to travel with me.
I have to say, this was the worst move I've ever made.
I opened the new shiny macbook and the first thing I noticed was that the internet seemed hit and miss. Moving around the room I managed to find a spot whereby pages woud load quickly. Strange, my £200 acer laptop was sh.t fast everywhere in the house. No matter, I packed it back away and carried on working on the Windows machine.
I've come to Newquay this week, and i've started to try and use the Macbook again in a hotel. The wireless signal in the room is low, and speedtest shows about 1 meg down and 3/4 meg up. That may sound bad to you, but remote desktop uses about 5k/sec (modem dial up speed) and the softphone, well, my Asterisk VOIP setup is confugured to use the GSM codec so that's 8k/sec each way + overheads.
This whole setup was deliberately designed to be "thin" so I can travel with ease and work on bad connections like USB internet sticks.
Anyhow, the Macbook was unable to hold a stable connection to the remote desktop or SIP phone, even though the speed test showed a whopping 1 meg up and down. What you may not be aware of is that there is more to a connection than the speed, there is the quality as well. How many packets are lost / how much "jitter" is on the line.
Anyhow, we're not living in the 3rd world, I ran a PINGTEST and it showed a small amount of jitter but told me the line was class B, online games may suffer but voip should be fine.
I unpacked the Acer, placed it in exactly the same spot as the Apple had sat in and it worked beautifully with 1 bar of wireless signal, all day long. Phone calls were clear.
So you know now what I'm thinking. I'm sitting here with my £200 acer because I can't use the £1500 macbook air 11. I paid nearly sixteen hundred pounds for this piece of .... and it doesn't ...... work. Time to contact Apple support.
2nd Mistake!
Representitive 1: - Told me that I can't compare the Macbook Air to the Acer, the Acer has Google Chrome and everybody knows Google Chrome is the fastest browser. I was told there was nothing more he could do, its probably a bad line at the hotel. When I explained the Acer works fine for voip I was told well, maybe it is getting a better signal. I explained the Acer has the cheapest possible parts inside it and paid 1500 for this macbook, expecting it to have quality parts inside and was told I'd paid for the size, because its so small but its not considered "powerful". Apple do you train your staff? Clearly not.
Thank god I wasn't paying to talk to this moron.
Representitive 2: - Had no idea what packet loss or Jitter was, got me to do a speedtest and said that looks fine. Then he got me to remove the WIFI adapter and re add it in the network settings.
Guess what, nothing changed, its still the same hardware and software.
Representitive 3: - Still not really understanding "quality" issues with the networking interface, I was asked to install the latest Java client. I did it, only because I wanted to comply with Apples wishes so they'd help me, but they weren't helping and Java has nothing to do with the network adapter, so that was useless advice too.
Apple seem to have no idea there is a problem, even though Google has pages and pages of people saying the same as me, and their own discussion forums have thousands of people complaining https://discussions.apple.com/thread/2664670?start=0&tstart=0
Finally, late yesterday whilst speaking to d.ck head number 3 at Apple support, we found a forum post talking about a fix, 10.7.1 update. I told d. head number 3 about the update and he suggested I applied it. So I did and everything looked great, for a whole evening.
This morning, I switched on again and the same thing, slow remote desktop, choppy unusable phone. Remember the phone needs less than 20k for a conversation, thats 0.2 meg. Speedtest again showing a whole meg both ways.
I called apple support again, this time being a little forceful, and I've asked for this to be escalated, but the bottom line is - they have no fix, they don't aknowledge this as a problem and I was told LION is new, so maybe it's got a bug..
I told the guy on the phone this is a network driver issue, the intermittency of the problem shows that and the Apple's lack of settings for the network adapter means the unit is autonegotiating with the router and choosing speed and duplex settings on its own. Sometimes it does that correctly, other times not and the connection although fast has a lot of noise / packet loss / corruption.
I've found a workaround, you put the unit to sleep and wake it up again and it runs fast until the next shutdown. Not really acceptable seeing as I was paying for "the cream of the crop".
I will definately not be recommending Apple products, and i'll certainly not be replacing the Windows laptops in my business with Apple's toytown system- i'd go out of business if I had to rely on this.
All there is left now, is to look at Boot camp and see if I can wipe this waste of space linux hack from the unit and install Windows 7.

Vendor Statistics Report ALV

Hi,
I am looking for a Vendor Statistics report developed by an ABAPER which basically shows how late a PO was, If full quantity was GR etc basically a Vendor Evaluation report,
Points will be awarded any Vendor Evaluation Type report will be helpful ALV format pref
Thanks
Adeel

Hi Adeel
you can get a start from this:
EKPO , JSTAT will be you might be helpful
Get the object number... for the order...
CALL FUNCTION 'STATUS_READ'
EXPORTING
client = sy-mandt
objnr = p_objnr
only_active = 'X'
TABLES
status = lt_status
EXCEPTIONS
object_not_found = 1
OTHERS = 2.
SELECT txt04 FROM tj02t
INTO lv_status
WHERE istat EQ wa_status-stat
AND spras EQ 'E'.
ENDSELECT.

What is "Default" in Usage Statistics report detail?

I am running RoboHelp Server 8 and am outputing the "Usage Statistics" report which has an option to report on Browsers and Operating Systems. In the results for both is a category called "Default" (see images below). Does anyone know what this means? If you can answer that, maybe you can also tell me what the "Unknown" operating system is? I suspect this maybe an Apple Mac user as I know one of our users has tried accessing the help through the application installed on a Mac.
Read the RoboColum(n) for a tips, tricks and musings on the Technical Communication Suite products.
Follow the RoboColum(n) on Twitter

Further to my previous post, I can confirm that the "Unknown" operating system is definitely NOT an Apple Macintosh user. Having run the report on another area, a seperate "Macintosh" category is displayed thus leaving me in the dark again. If anyone can shed some light on this and the "Default" category I'd be very grateful.
Read the RoboColum(n) for a tips, tricks and musings on the Technical Communication Suite products.
Follow the RoboColum(n) on Twitter

Report Definition not found - calling a report from JDev.

Hello,
I have the following on my local machine:
BI Publisher
Jdeveloper
I created a report in BI publisher called Test1, and now I am creating a web service in jdeveloper to call it.
I am having issues with calling it from my jdeveloper.
here is the code I am using to call my report (below).
Since BI publisher is installed on my local machine I tried both paths but neither are working
1- final String reportAbsolutePath = "C:\\Test1.xdo";
2 - "http://aramamni-us.us.oracle.com:9704/xmlpserver/~administrator/Learn/Test1/Test1.xdo"
I am sure 2 will not work since System.out.println("calling " + myPort.getEndpoint()); returns
"calling http://localhost:9704/xmlpserver/services/PublicReportService"
I know the problem is with this line of code but I can't figure it out.
repResponse = myPort.runReport(repRequest,username,password);
    public static void main(String[] args) {
        try {
            bip_webservice.proxy.PublicReportServiceClient myPort = new bip_webservice.proxy.PublicReportServiceClient();
            System.out.println("calling " + myPort.getEndpoint());
            // Add your own code here
             final String username = "Administrator";
             final String password = "Oracle123";
             //final String reportAbsolutePath = "/Web Service Test/HR Manager/Employee Salary Report.xdo";
             final String reportAbsolutePath = "C:\\Test1.xdo";
            // Testing runReport
            System.out.println("Testing runReport Service");
            ReportRequest repRequest = new ReportRequest();
            repRequest.setReportAbsolutePath(reportAbsolutePath);
            repRequest.setAttributeTemplate("default");
            repRequest.setAttributeFormat("pdf");
            repRequest.setAttributeLocale("en-US");
            repRequest.setSizeOfDataChunkDownload(-1);
            System.out.println("1111111111111");
            ParamNameValue[] paramNameValue = new ParamNameValue[1];
            paramNameValue[0] = new ParamNameValue();
            paramNameValue[0].setName("EMP");
            System.out.println("2222222222");
            paramNameValue[0].setValues(new String[] {"20"});
            //paramNameValue[1] = new ParamNameValue();
            //paramNameValue[1].setName("p_job");
            //paramNameValue[1].setValues(new String[] {"CLERK"});
            repRequest.setParameterNameValues(paramNameValue);
            System.out.println("3333333333333");
            ReportResponse repResponse = new ReportResponse();
            System.out.println("4444444444");
            System.out.println(username);
            System.out.println(password);
            System.out.println(repRequest);
            repResponse = myPort.runReport(repRequest,username,password);
            System.out.println("55555555");
            String contentType = repResponse.getReportContentType();
            System.out.println(contentType);
            byte[] baReport = repResponse.getReportBytes();
            FileOutputStream fio = new FileOutputStream("C:\\temp\\test.pdf");
            fio.write(baReport);
            fio.close();
        } catch (Exception ex) {
            ex.printStackTrace();
    }Output -> Report Definition not found
calling http://localhost:9704/xmlpserver/services/PublicReportService
Testing runReport Service
1111111111111
2222222222
3333333333333
4444444444
Administrator
Oracle123
bip_webservice.proxy.ReportRequest@8238f4
WARNING: The received SOAP fault contains non standard fault element: "{http://xml.apache.org/axis/}hostname". This element will be ignored.
javax.xml.rpc.soap.SOAPFaultException: oracle.apps.xdo.webservice.exception.OperationFailedException: PublicReportService::generateReport Failure: Due to oracle.apps.xdo.servlet.CreateException: Report definition not found:C:\Test1.xdo
     at oracle.j2ee.ws.client.StreamingSender._raiseFault(StreamingSender.java:565)
     at oracle.j2ee.ws.client.StreamingSender._sendImpl(StreamingSender.java:400)
     at oracle.j2ee.ws.client.StreamingSender._send(StreamingSender.java:113)
     at bip_webservice.proxy.runtime.PublicReportServiceSoapBinding_Stub.runReport(PublicReportServiceSoapBinding_Stub.java:3733)
     at bip_webservice.proxy.PublicReportServiceClient.runReport(PublicReportServiceClient.java:173)
     at bip_webservice.proxy.PublicReportServiceClient.main(PublicReportServiceClient.java:84)
Process exited with exit code 0.Thanks for the help - what can i do to solve this.
Edited by: Rooney on May 25, 2010 2:57 PM

I tried to follow the suggestions provided in the earlier post and called the BI report from SOAP UI
but i am getting an error
oracle.xdo.webservice.exception.OperationFailedException: PublicReportService::generateReport for reportAbsolutePath [/users/weblogic/Sample/Training.xdo] failed: due to oracle.xdo.servlet.CreateException: Report definition not found:/users/weblogic/Sample/Training.xdo [username=weblogic]
I am not sure what is the issue.
Can you please guide me on the same?
Regards
AR

Lync 2013 Call Quality Reporting Time Intervals

Does anyone know the time interval used to report the metrics associated with the Lync 2013 call quality reports? If so, is there documentation that confirms that?
For example, Max Packet Loss Rate gives you a percentage. However, I haven't found any documentation that says how long a sample period is. Was that Max Packet Loss rate experienced for, at least, 1 second, 5 seconds, 30 seconds, 1 mintue etc?
I haven't found any conclusive information regarding how long of a period they may have experienced that max packet loss number, nor at what point during the call.
Thanks in advance for the help!

Hi,
The network health is expressed by the number of poor audio streams, where a poor stream is defined as having a packet loss rate over 1 percent and a maximum packet loss rate over 5 percent.
You can refer to the following two links to understand it easliy:
http://blogs.technet.com/b/jenstr/archive/2013/06/11/viewing-network-health-trends-using-qoe-data.aspx
http://blogs.technet.com/b/nexthop/archive/2012/12/10/troubleshooting-call-quality-locally-with-snooper.aspx
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Looking for _synchronous_ call to report

Hello,
(Oracle 9iR2-902-EE)
I have the following sequence:
--plsql_batch1
--report1
--plsql_batch2
batch2 starts only after report completed.
I am aware of
--UT_HTTP package
--SRW package
--HOST procedure
Looks like all of possibilities are non-blocking
asynchronous calls. Some forum posts describe
checking on status (SRW package) in loop.
Is there some better solutions to call report
not from URL but more "natively" with better control
over end time and status (blocking mode).
Thank you in advance for any advice.
Alex V.

Hi,
In your case, you can use the following cmdlet to determine unused mailboxes. If a mailbox has never logged on, the Lastlogontime is empty.
Get-Mailbox|Get-MailboxStatistics |fl DisplayName,Lastlogontime
Use the cmdlet below to get publicfolder's last access time.
Get-PublicFolderStatistics |fl name,LastAccessTime
Get all distribution lists used over last 30 days.
Get-MessageTrackingLog -EventId Expand -ResultSize Unlimited |Sort-Object RelatedRecipientAddress | Group-Object RelatedRecipientAddress |Sort-Object Name | Select-Object @{label=”PrimarySmtpAddress”;expression={$_.Name}}, Count
Hope this can be helpful to you.
Best regards,
Amy Wang
TechNet Community Support

Error:"non-MTOM endpoint received a MTOM message"

Hello,
Appreciate someone could show right points toward below issue:
Enviornment: Oracle Weblogic Server /portal10.3.0
1) We are using Invoking a webservice which is sending some binary content in SOAP 1.2 MTOM
2) On the basis of published wsdl , I built a own Weblogic ServiceControl (com.bea.controls.ServiceControl) class ...
3) I could print the response from the web service but when the actual message is decoded to response object...i thinks it's failing...
weblogic.wsee.codec.CodecException: non-MTOM endpoint received a MTOM message"
at weblogic.wsee.codec.soap11.SoapCodec.decode (SoapCodec.java:170)
at weblogic.wsee.ws.dispatch.client.CodecHandler.decodeOutput(CodecHandler.java:127)
Regards

Can you test back-end MTOM end point through a route node and see if its working? I doubt the issue has nothing to do with SC but with the MTOM end-point implementation.
non-MTOM endpoint received a MTOM message") error: hints that the service call out is sending a mtom message but the end-point is not accepting the message
Manoj
Edited by: mneelapu on Jun 8, 2009 11:25 AM

Non-Immersive Endpoints Call Statistics Reports - jitter, packet loss

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