[Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx
Jonathan Hunter
hunterj91 at hotmail.com
Wed May 31 19:24:27 UTC 2017
Hi Guys,
Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with?
I understand mos and quality percentage but what other values are a good indicator?
Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented.
Many thanks
Jon
"variable_rtp_audio_recv_pt": "8",
"variable_rtp_audio_in_raw_bytes": "374960",
"variable_rtp_audio_in_media_bytes": "374616",
"variable_rtp_audio_in_packet_count": "2180",
"variable_rtp_audio_in_media_packet_count": "2178",
"variable_rtp_audio_in_skip_packet_count": "92",
"variable_rtp_audio_in_jitter_packet_count": "0",
"variable_rtp_audio_in_dtmf_packet_count": "0",
"variable_rtp_audio_in_cng_packet_count": "0",
"variable_rtp_audio_in_flush_packet_count": "2",
"variable_rtp_audio_in_largest_jb_size": "0",
"variable_rtp_audio_in_jitter_min_variance": "28.57",
"variable_rtp_audio_in_jitter_max_variance": "116.33",
"variable_rtp_audio_in_jitter_loss_rate": "0.02",
"variable_rtp_audio_in_jitter_burst_rate": "0.98",
"variable_rtp_audio_in_mean_interval": "20.39",
"variable_rtp_audio_in_flaw_total": "43",
"variable_rtp_audio_in_quality_percentage": "97.00",
"variable_rtp_audio_in_mos": "4.47",
"variable_rtp_audio_out_raw_bytes": "210012",
"variable_rtp_audio_out_media_bytes": "210012",
"variable_rtp_audio_out_packet_count": "1221",
"variable_rtp_audio_out_media_packet_count": "1221",
"variable_rtp_audio_out_skip_packet_count": "0",
"variable_rtp_audio_out_dtmf_packet_count": "0",
"variable_rtp_audio_out_cng_packet_count": "0",
"variable_rtp_audio_rtcp_packet_count": "0",
"variable_rtp_audio_rtcp_octet_count": "0",
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