Packet lost

We have an issue with a voice gateway. The scenario is:
GW3660 ----> ADSL -----> 7200 -----> internet
When I put the "sh call active voice brief" during a call I show:
20CA : 8828 1781354800ms.1 +3020 pid:7 Originate 09538xxxx active
dur 00:04:30 tx:5626/69666 rx:10695/180544
IP x.x.19x.x:17424 SRTP: off rtt:0ms pl:169880/3010ms lost:95/41/178 delay:80/60/150ms g729br8
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
Telephony call-legs: 1
SIP call-legs: 1
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
When I heard the voice lost during the packet lost number increases but I deploy the QoS in the link ATM between GW and 7200.
I insert the tx-ring at 3, fragment delay on PPP at 4 (about 512 byte) and cRTP and LLQ in outbound of 3660.
Now I don't what anything else configure to resolve the issue.
Does anyone know how?
Thank you.
Paolo

Hi,
Here we are the QoS configuration in 3660:
interface ATM3/0
bandwidth 1024
ip address x1x.3x.1xx.6 255.255.255.252
ip nat outside
ip virtual-reassembly
no atm ilmi-keepalive
scrambling-payload
impedance 120-ohm
service-policy output VOIP
pvc 10/100
protocol ip x1x.3x.1xx.x broadcast
tx-ring-limit 3
oam-pvc 0
encapsulation aal5snap
protocol ppp Virtual-Template1
interface Virtual-Template1
bandwidth 1024
ip unnumbered ATM3/0
no ip route-cache cef
ip tcp header-compression iphc-format
load-interval 30
ppp multilink
ppp multilink fragment delay 4
ppp multilink interleave
max-reserved-bandwidth 50
ip rtp header-compression iphc-format
class-map match-any voice-control
match ip dscp cs3
match ip dscp af31
class-map match-any voice
match ip dscp ef
policy-map VOIP
class voice-control
bandwidth percent 2
class voice
priority 256
class class-default
fair-queue
In 7200 there is the same configuration except the policy VOIP in output interface because there some bugs that make instable the router.
Could it be that?
Thanks.
regards,Paolo

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This is the actual screen on the station on one of the sites. The 192.168.1.254 is the remote site inside interface address. The Maximun Size without Framenting is 1370 but still I got some packet loss.
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Hi,
I need your help to explain me the next behavior.
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Rick,
Searching in Internet and Riverstone, we found the next document: Ping rate-limiting feature added in 7..x and newer
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Before:
PE_Mty#ping vrf SONUS-VOIP
Protocol [ip]:
Target IP address: 192.168.57.2
Repeat count [5]: 1000
Datagram size [100]:
Timeout in seconds [2]:
Extended commands [n]:
Sweep range of sizes [n]:
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PE_Mty#
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Protocol [ip]:
Target IP address: 192.168.57.2
Repeat count [5]: 1000
Datagram size [100]:
Timeout in seconds [2]:
Extended commands [n]:
Sweep range of sizes [n]:
Type escape sequence to abort.
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Recently,I capture some RTP packets when user meeting the dialogue jitter and latency.
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Probably you have captured RTP traffic near the voice gateway. Right?
I suppose that you have bidirectional packet loss.
In this case, if you use wireshark near one of the sources you can observe only one direction loss.
You must try to use wireshark in both network segments.
P.S. in the bottom of the your image we can see ip addresses of your net.
Regards.

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Hi,
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http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/wrp400/release/notes/WRP400_RN_v2-00-26.pdf
Additional Provisioning Parameters
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NOTE
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Use the syntax shown in the following example, and described below.
Example:
softirq_active_voice=0,softirq_polling_cnt=1
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•softirq_polling_cnt: Specify the polling count. Valid values are integers from 0 to 10.
After upgrade, default parameters fixed my cutoff issue.
Regards.

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This question was solved.
View Solution.

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Your system: Windows XP version 5.1
Java version: 1.7.0_04 (x86)
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If your network is slow, there's got to be a specific reason. We can probably help you isolate the various causes and find out what is causing it to behave so slowly.
I do not know what level of expertise you have when working with computers or networks so I will try to be specific. If we are to help you, you will need to be specific and accurate when responding. Vague answers from you will only net vague guesses from us. Specific answers from you will likely result in our being able to provide you with specific suggestions and fixes.
If the Mac is fast when running Windows, but slow when running OS X then it's not likely that you actually have a hardware defect (bad network interface, disconnected wifi antenna, etc. as these problems would manifest whether running OS X or Windows on the same hardware.
First, tell us about your Mac and your network. Which Mac model do you have? How much memory does it have? What OS version is it running? If you run "Software Update" is this Mac fully and completely up-to-date with the latest patches?
A long while back there was a known bug in Leopard that would cause a MacBook Pro to be slow if using a wireless network while on battery power (it would be fast if connected to AC power). That was patched long ago. (hence my interest in which OS revision you are running and whether it is fully patched)
What brand & model network router are you using?
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In "System Preferences" -> "Network", select your active network interface (typically either "Ethernet" or "Airport"), then select the "Advanced..." button in the lower right corner. On the next select the "Ethernet" tab across the top (usually the far-right). Is "Configure:" set to "Automatically"? Is the "MTU:" show a grayed out value that reads "Standard (1500)"? If not, what values do you see?
Next
Start "Network Utility" (it's under "Applications" -> "Utilities"), go to the "Info" tab, select your active network interface (probably ether "Ethernet (en0)" or "Airport (en1)").
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Next
I'd like to isolate and test the quality of the connection between your Mac and your router (ignoring the rest of the Internet). If the connection between those two components is solid and healthy, then any slow down would be a problem elsewhere in the network and not, specifically, your Mac.
There is something called "MTU" (maximum transmission unit) that I wont get into just now. A problem with MTU can cause slow delivery of packets on what should be a blazingly fast network in otherwise excellent health, but I'll set that aside for now. 99.9% of the time you should never attempt to alter the MTU.
Go into "System Preferences" -> "Network", select your network interface (either Ethernet or Airport -- whichever you happen to be using) then note the value displayed for "Router:"
Back in "Network Utility", select the "Ping" tab and type in the value displayed for your Router (which you noted above) into the box labeled "Enter the network address to ping." Usually this is a number that looks like "192.168.1.1" or "192.168.0.1" or "10.0.0.1", etc. (almost always ends with a ".1". After entering the number, verify that you have select "Send only [10] pings" (that's the default), then click "Ping" to start the test.
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Your answers to these questions will help us rule IN or rule OUT various possible causes.

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Please someone help in capturing the live streaming in a file using JMF......
Thanks..

Hi, I have a problem with RTPExport output video files. One side streams H263/RTP(AVTransmit2.java) and other write this steam to a file by RTPExport.java. When network conditions are ideal, output video file has same fps and same number of frames like original file. Problem occures, when theres packet lost in network, then output file has different fps,and also has less frames like original video(because it didnt write missing frames to file, and thats why it get shorter). Pls how can I achieve output file that will have the same fps like original one? How to write to file an identical copy of what I can see while receiveing video with AVReceive2.java? Its there a way to modifi rtpexport or avreceiver to do this? Thanks!

Problem with RTPExport output video files

Hi, I have a problem with RTPExport output video files. One side streams H263/RTP(AVTransmit2.java) and other write this steam to a file by RTPExport.java. When network conditions are ideal, output video file has same fps and same number of frames like original file. Problem occures, when theres packet lost in network, then output file has different fps,and also has less frames like original video(because it didnt write missing frames to file, and thats why it get shorter). Pls how can I achieve output file that will have the same fps like original one? How to write to file an identical copy of what I can see while receiveing video with AVReceive2.java? Its there a way to modifi rtpexport or avreceiver to do this? Thanks a lot!

Trubka wrote:
When network conditions are ideal, output video file has same fps and same number of frames like original file. Problem occures, when theres packet lost in network, then output file has different fps,and also has less frames like original video(because it didnt write missing frames to file, and thats why it get shorter). Okay, first off, the second file is smaller on purpose. RTP intentionally drops packets that are old/out of order in order to make sure real-time video stays as close to real time as it can. This is by design, so there's really nothing that can be done about it.
How to write to file an identical copy of what I can see while receiveing video with AVReceive2.java? Technically speaking, what you're getting in the RTPExport is exactly what you got on the receiving end. Any frames that are dropped during transmission will not be seen by the receiver, nor saved by the receiver.
Pls how can I achieve output file that will have the same fps like original one? I'm not 100% sure that you can, but, you can give the following idea a try. I make no guarentees that it'll work, but it should work for you...
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/RTPConnector.html]
That example is an example of a "custom transport layer" for RTP connections. Essentially, it's some code that's handed the RTP packets on the transmission end, and it's expected to deliver those RTP packets on the other end. And it doesn't care about how they get from A to B, only that they do.
If you were to replace the UDP socket in that example with a TCP socket, you would be guarenteed not to drop packets due to network reasons. Every RTP packet you were handed by the transmitter, you would then hand to the receiver. There is no guarentee that none of the packets would be cast away as "old" by the RTP protocol itself, but there's also no guarentee any of them would be. It's a crap-shoot at best, but it's certainly worth a try.

Packet lost

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