====== Call log ======
The call log lists SIP calls made through Internet Gate.

:!: Call logging is by default disabled. It can be enabled on the [[log_configuration_page#Call Log|Log Configuration]] page, where you also can decide if you additionally want to collect call quality related metrics, send it to a syslog server, etc.

===== Typical entries =====
A SIP call will usually generate the following call records in the call log:

  Ringing start for [sip:1234@domainx.com 111.111.111.111] -> [sip:5678@domainy 222.222.222.222]

  * 1234@domainx.com is the calling parties SIP address, 111.111.111.111 is the calling clients IP address.
  * 5678@domainy.com is the called parties SIP address, 222.222.222.222 is the called clients IP address.
There is only one Ringing start record for a call, even if this call rings several parallel because of call forking.

  Start of session M2Qyg67bYjIm: [sip:1234@domainx.com 111.111.111.111] -> [sip:5678@domainy 222.222.222.222]
The Start of session record is created when the call is answered by the called party. //M2Qyg67bYjIm// is a unique session ID for this call. It is composed of the beginning of the incoming call ID, of the end of the outgoing call ID, and of a hash value of the remaining part of the involved call IDs. The incoming and outgoing call ID will usually be the same, except for special scenarios, e.g. when a back-to-back SIP user agent is involved.

  End of session M2Qyg67bYjIm: [sip:1234@domainx.com 111.111.111.111] -> [sip:5678@domainy 222.222.222.222]
  Duration: 2 sec, Terminate reason: User Request
The End of session record is created when the call is finished. Terminate reason User Request is the regular case, Timeout is an exceptional case if one party of the call does not respond any more.

  > Bytes: 11200, Pkts: 56, missing: 0 (max 0), reo.: 0, Jitter avg: 0 ms (max: 7 ms), last PT: 0 (PCMU), CN: no, MOS: 4.40
  < Bytes: 13400, Pkts: 67, missing: 0 (max 0), reo.: 0, Jitter avg: 0 ms (max: 0 ms), last PT: 0 (PCMU), CN: no, MOS: 4.40
These two records follow immediately after the End of session record in the log. They are only generated when //including Media statistics// is set up on the [[log_configuration_page#Call Log|Log Configuration]] page for call logging. They contain results of the RTP packet monitoring which is performed for the RTP streams belonging to this call.
The first line contains information about the RTP packet stream from the calling to the called party, the second line contains information about the opposite direction.
  * //Bytes// is the number of bytes (including IP header);
  * //Pkts.// is the number of RTP packets. 
  * //missing// tells the number of missing (lost) packets. 
  * //max// is the number of continuously missing packets, so e.g. a value of 3 means that e.g. the packets with sequence numbers 3001,3002 and 3003 were missing. 
  * //reo// is the number of reordered packets, so packets which were not received in the normal order. 
  * //Jitter avg.// is the average Jitter value in milliseconds over the total call duration. [[wp>Jitter]] 
  * //max// is the maximum jitter value during the call. 
  * //Last PT// is the last packet's payload type value from its RTP header. 
  * //PCMU// here tells that the used codec in the last RTP packet was PCMU in this example. 
  * //CN// is either yes or no and tells if Comfort Noise packets (used for silence compression) were present during the call. [[wp>Comfort_noise|Comfort noise]]
  * //MOS// means "Mean opinion score"; it is a value between 1 and 5 and gives an estimation how listeners might experience this call, with 1 being "bad" and 5 being "excellent". This value is calculated mostly based on the used codec, and on the packet loss distribution during the call. Note that this value does not take into account possible packet loss or other problems happening to the packet stream after it has traversed this box. [[wp>Mean_opinion_score|MOS]]

  RTCP results from called / calling party: missing packets 1 / 1, jitter 3.0 / 2.1 ms (max 8.8 / 6.1 ms); total round trip delay 20.6 ms
This record contains information which is retrieved from RTCP messages which are sent between the two endpoints of the call. 
It is only generated when //including Media statistics// is set up on the [[log_configuration_page#Call Log|Log Configuration]] page for call logging, and if there was RTCP signaling at all between the two endpoints.

Except for the total round trip delay, the information comes in pairs, the first number telling information which is reported via RTCP from the called party, the second value from the calling party. '**-**' is displayed if the corresponding value is not available, e.g. because an endpoint did not send any RTCP messages.

  * //missing packets// how many packets were missing (not received) by the reporting endpoint.
  * //jitter// average of the jitter values which were reported by the endpoint. [[wp>Jitter]]  
  * //max// the biggest jitter value which was reported by the endpoint. 
  * //total round trip delay// total round trip delay between the two endpoints; this is the average time it took for RTCP packets to get from one endpoint to the other and back.


===== Severity levels =====
The ringing start message has severity level //debug//.\\
The start and end of session messages have severity levels //notice//.\\
The media statistics messages have severity levels //info//.\\
There might also be //error// messages logged if any errors occur.\\

By selecting a suitable level for filtering on e.g. a syslog server you can select the level of information you save.
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====== ======
Read more: [[[[sip:start|SIP]] [[wp>Real-time_Transport_Protocol|RTP]] [[wp>RTCP]]