we have on our networks ponctual voice quality degradation for our ToIP user; we have noticed some paquet lost between gateway and IP Phone, but still don't know where the packet are lost.
In order to perform easily tests, I am looking for a tool able to generate RTP traffic, with a client and a server (ideally, with the possibility to forge some fileds like CoS), permitting to measure the quantity of lost packets.
Thanks in adance,
I don't know an RTP traffic generator. But, in these cases, I capture the traffic and I analyze it with wireshark. It's free but very complete and helpful.
To get the traffic, I use tcpdump on linux server, wireshark with winpcap on windows server or span-port (monitor session port) on network switch.
In this way you can check the real situation.
Wireshark has a graphical RTP statistic and analysis tool and a graphical VoIP session tool. You can also play RTP stream.
More info here:
You could try using dos ping
ping -v 184 -l 170 REMOTE IP ADD
This will generate IP packet (ICMP) using DSCP 46 (EF) and a packets size if 170 bytes
Hope this helps
here is a good tools to have some statistics and monitoring
thanks for your answers. Here what I have finally done:
- I configured the IP SLA to monitor UDP jitter and lost packets. I configured the IP SLA to simulate G711 voice flow. It permit to generate regularly the equivalent of a20 seconds conversation.
- I use a tool to graph IP SLA result (jitter, lost packets)
- Wireshark to monitor packets on some points of the network.