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RTP packets stop randomly


(Bert) #1

Been having issues with random calls where during call the person on the other end can’t hear us but we can always hear them.If the person doesn’t hang up the audio comes back after 30 to 40 seconds.This can happen to any call at random the only thing consistent is we can always hear them. Our sip provider monitored for a while and said that the RTP packets from our end just stops. Has anyone had this issue. Will take any suggestion on what to monitor or check to clear this up.
FreePBX 13.0.194.5


#2

Use tcpdump to capture all traffic on the PBX. Move the capture to your workstation and examine with Wireshark. If the RTP indeed stops flowing to the trunk, see whether it stopped coming in from the extension.

Continues flowing to trunk: Problem is with your network (firewall, router, modem, etc.) or ISP. Try to capture traffic at the modem to troubleshoot further.

Stops flowing to trunk but is still coming in from extension: Problem is with Asterisk, but I’ve never seen it before. Has this occurred when multiple calls were in progress? If so, were they all affected at the same time?

Stops coming in from extension: If local, problem with LAN or device. If remote (VPN, mobile data, etc.) please provide details.

I run this as root in the background:
tcpdump -s 0 -C 100 -W 100 -w rbuf -Z root &
It creates rbuf00 … rbuf99, each 100 MB, overwriting the oldest (10 GB total space needed). After the trouble occurs, stop the tcpdump, locate the relevant file or files (compare CDR timestamps with capture timestamps) and move them to your workstation for analysis.


(Bert) #3

Thanks for pointing me in a direction, it is appreciated