is it possible to check how long was a VoIP call, directly through ser or some ser tool?
thanks Giuseppe
if you use the acc module to account the SIP messages, then you get records for INVITE and BYE which are the start and the end of a call. For more details see the acc module:
http://openser.org/docs/modules/1.1.x/acc.html
Cheers, Daniel
On 05/09/2006 11:52 AM, Giuseppe Parlato wrote:
is it possible to check how long was a VoIP call, directly through ser or some ser tool?
thanks Giuseppe
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Hi Daniel,
If I record INVITE and BYE then the SER becomes sessions-stateful (as mentioned) in the docs. There is also mentioned, that this is very bad for performance. Is it better to do the accounting on the RTP Proxy? Is this what the docs say? Or did I missunderstod something?
chris...
Daniel-Constantin Mierla wrote:
if you use the acc module to account the SIP messages, then you get records for INVITE and BYE which are the start and the end of a call. For more details see the acc module:
http://openser.org/docs/modules/1.1.x/acc.html
Cheers, Daniel
On 05/09/2006 11:52 AM, Giuseppe Parlato wrote:
is it possible to check how long was a VoIP call, directly through ser or some ser tool?
thanks Giuseppe
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Christoph Fürstaller wrote:
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Hi Daniel,
If I record INVITE and BYE then the SER becomes sessions-stateful (as mentioned) in the docs. There is also mentioned, that this is very bad for performance.
Just because you use record_route and account the INVITE and BYE requests does not make (open)ser call stateful (or dialog-stateful in SIP terms). E.g. if the BYE request will never be sent (e.g. both clients crashes) openser will not care about it. Thus you do not get a BYE CDR in your accounting table.
Is it better to do the accounting on the RTP Proxy? Is this what the docs say? Or did I missunderstod something?
Doing accounting on a node which stays within the media session (e.g. a gateway, RTP proxy, B2BUA ...) is much more accurate. Thus, using the logs from the RTP proxy is much more accurate and prevents you from "dead" session as the RTP proxy will drop down the session after some time of inactivity and will terminate the session in the logs (CDR). Nevertheless this requires to proxy each call which will lead to more traffic on your RTP proxy and usually bigger round-trip-times.
regards klaus
chris...
Daniel-Constantin Mierla wrote:
if you use the acc module to account the SIP messages, then you get records for INVITE and BYE which are the start and the end of a call. For more details see the acc module:
http://openser.org/docs/modules/1.1.x/acc.html
Cheers, Daniel
On 05/09/2006 11:52 AM, Giuseppe Parlato wrote:
is it possible to check how long was a VoIP call, directly through ser or some ser tool?
thanks Giuseppe
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
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