Debug Strategy For One-way Audio Calls

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Asterisk Users 5 Comments

Hello everybody,

from time to time, we get so-called simplex / one-way audio calls, where one party cannot hear the other. The only thing in common is that is does happen with calls via SIP trunk, not ISDN and not internal calls. Nothing strange in verbose and SIP logs. Could even be some weird intermittent firewall issue I guess.

Apart from logging all traffic 24/7 via tcpdump (not really convenient), can you give me some ideas how to debug this kind of issue?

Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.

5 thoughts on - Debug Strategy For One-way Audio Calls

  • 2013-05-02 13:19, Marie Fischer skrev:

    Voipmonitor.org is great for debugging voip. You can either use only the sniffer (opensource) and use mysql + the pcap files or you can also buy the commercial webgui. Either way, it’s a great product.

    /Johan

  • @Marrie For one way audio as a debug strategy you can enable RTP debug and see whether you have both way packets flow SENT and GOT.

    Regards

  • Not sure about older calls, but outbound was missing the last few times. We use call recording via MixMonitor and the recording has both flows, so I guess rtp debug would have shown both as well.