Read DTMF D
I am using Connect 8.1.2 with an FMG on its own virtualized server. When I open a meeting and connect to a PGi conference bridge using Universal Voice, I sometimes see bursts of dropped packets in the FMG core.log such as the following:
2012-02-08::08:56:10.922 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=1 Video=0
2012-02-08::08:56:10.941 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:10.962 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:10.981 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:11.002 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=1 Video=0
2012-02-08::08:56:11.002 DEBUG CALLLEG 2920 [LEG ID:713] - Read DTMF D
2012-02-08::08:56:11.103 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:11.124 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:11.144 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:11.163 DEBUG SIP 2920 [LEG ID:713] - Number of media packets dropped: Audio=2 Video=0
2012-02-08::08:56:11.404 DEBUG CALLLEG 2920 [LEG ID:713] - Read DTMF D
I am connecting to the phone conference with an office phone to use as audio input to evaluate audio quality. I experience moments in meetings where the audio "cuts-out" for more than 500ms, but have not determined fully whether the dropped audio media packets are the culprit.
Does anyone know:
-why the FMG would be receiving a DTMF D from the conference call, especially after multiple dropped packets in a row? Is this just provider specific?
-why the FMG would be dropping packets (it is under very low call volume and it is co-located in the same data center as the FMS cluster and in the same region as the SIP trunk provider)?
-is there any test data validating the claim that the FMG can handle 100 concurrent calls for every 2 CPU cores? It seems with the transcoding that must be done (G711u <--> speex/NM22k) that this is an optimistic number.
-what is the usual bottleneck/main hurdle when using the FMG for Universal Voice (CPU performance, network latency, audio input source, codecs used, ...)?
Thanks so much for any help or insights that can be provided.
Best regards,
Jagat
