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CDR ‐ Call Detail Record
Minh edited this page Nov 20, 2023
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When you make a phone call to another phone number, from the perspective of the LibreSBC, there are 2 parts/legs of the call session:
- Ingress (aka A leg or the incoming call leg) to the LibreSBC is the incoming connection from the originator (caller)
- Egress (aka B leg the outgoing call leg) from the LibreSBC is the outbound connection to the recipient of the call (callee)
Call Detail Records are the data recorded during each call session, each leg will have it own CDR.
Parameter | Description |
---|---|
uuid | CDR uuid, unique per leg |
seshid | Session id, unique per call session |
direction | Call direction, inbound outbound
|
sipprofile | SIP profile name where call was engaged |
context | Context of sip profile |
nodeid | SBC node id where call was processed |
intconname | Interconnection name |
gateway | Gateway name, available for outbound direction only |
user_agent | SIP user agent, derived from SIP user agent header |
callid | SIP Call ID, derived from SIP call id header |
caller_name | Caller name |
caller_number | Caller number |
destination_number | Destination number |
start_time | epochtime when call was started |
answer_time | epochtime when call was answered |
progress_time | epochtime when 180 Ringing was process |
progress_media_time | epochtime when first media was process (it can be 183 early media or 200 OK) |
end_time | epochtime when call was hanged up |
duration | call duration in second, end_time - answer_time |
sip_network_ip | IP address of interconnection via SIP protocol |
sip_network_port | Port number of interconnection via SIP protocol |
sip_local_network_add | IP address of SBC via SIP connection |
transport | Transport protocol udp tcp tls
|
remote_media_ip | IP address of interconnection via RTP protocol |
remote_media_port | Port number of interconnection via RTP protocol |
local_media_ip | IP address of SBC via RTP connection |
local_media_port | Port number of SBC via RTP protocol |
read_codec | Media codec used by interconnection |
write_codec | Media codec used by SBC |
rtp_crypto | Media encryption algo used for SRTP |
hangup_cause | Cause why call was released |
sip_resp_code | SIP final response code, eg: sip:487 |
disposition | Who hangup the call |
status | alias of sip_resp_code
|
How to measure call metric?
- billable duration =
end_time
-answer_time
(duration) - waiting time for answering =
answer_time
-start_time
- waiting time for ringing =
progress_time
-start_time
orprogress_media_time
-start_time
(if progress_time=0) - if call answered: ringing_duration =
answer_time
-progress_time
oranswer_time
-progress_media_time
(if progress_time=0) - if call unanswered: ringing_duration =
end_time
-progress_time
orend_time
-progress_media_time
(if progress_time=0)