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We’ve been working with a customer using the AVM Fritzbox (or “Fritz!box”) this week trying to help get their SIP trunks up and running and get the call quality up to scratch…. not all CPE do VoIP and SIP the same way!!
With the Fritzbox we see a few problems including inconsistencies in downstream call quality that result from dropped packets and packets with out of sequence or missing time stamps.
At our end, we can capture at packet level and replay the media stream using the Telephony VoIP tools in Wireshark. Our network providers get similar tools straight off the Acmepacket (the session border controller), so we can work together to figure the problem out.
It can be very difficult for the customer to see or really unerstand whats going on, but these simple tools really help – we don’t have to explain most of the technical aspects, just let them hear the stream from their side.
We can also see these RTP details visually with the RTP stream analysis – problem packets are highlighted and summary stats provided.

A few practical pointers …
- Get pcap captures from yours or your customers network, or direct from your voice providers SBC (session border controller) – you’ll need access to a span/mirror port on the local network if you’re going to capture locally.
- Look at the SIP conversation graphically using the Wireshark Statistics > Flow Graph option – you’ll be able to see whether there are any obvious SIP problems.
- Make use of the SIP / RTP stats (look for X-RTP-Stat payload) – depending on what extensions are supported you’ll get various summary stats. {Use the RTP stream analysis in wireshark for similar purpose http://wiki.wireshark.org/RTP_statistics )
- You can replay the actual call very easily and see a graph of the audio if you use the Telephony menu > VoIP calls option. Again, this will give you a quick way of visually seeing whether the audio stream looks flakey.
- If you want to demonstrate to your customer, consider recording the up and downstream media streams separately to compare them. You can do this direct from Wireshark (RTP Stream analysis > Save payload) or with an audio capture tool like Audacity. Then you can record the audio as a wav file and no need for your customer to use Wireshark.