Hi all,

I've just released jSIPp 0.0.5. This release adds the "-rtp_sink"
command-line option. When this option is set, each call opens an
associated UDP socket, processes RTP packets coming in on that port, and
makes statistics on jitter, packet loss and out-of-sequence packets
available over the ZeroMQ statistics interface. The port of the UDP
socket is available in the [media_port] keyword for sending in the SDP.

Currently, these stats are published every 250ms for each call - I can
add an option in future to tune this frequency, if that would be
useful. The stats are documented at
https://github.com/rkday/jsipp/wiki/ZeroMQ#rtcp.

This support is still pretty limited - it can't send its own RTP packets yet, it
doesn't handle RTCP (either sending or receiving - so, for example, it
can't calculate round-trip time) and there's a hardcoded assumption that
the RTP profile is 8000Hz when calculating RTP timestamps. Nonetheless,
this is a feature that I've had quite a few requests for in SIPp, so I
hope it should make RTP performance testing easier. I intend to keep
working on it, based on any feedback I get about what would be the most
useful features.

The implementation should be quite efficient (in particular, all RTP
packets are handled on a dedicated thread using epoll/kqueue for
networking if available - it doesn't start a separate thread for each
RTP connection). That said, I haven't done any high-load testing at this
point - if you do, I'd be keen to get feedback on how high it can go.

This release is available from
https://github.com/rkday/jsipp/releases/tag/v0.0.5. To demo this new
function, I've also
released a new version of the web UI which shows the recent jitter,
packet loss and out-of-sequence packet rates averaged across all calls,
plus the jitter high-water-mark - that is available from
https://github.com/rkday/jsipp-web-ui/releases/tag/v0.2.0.

Feedback or constructive criticism, as always, is very welcome!

Best,
Rob

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