Hi Aqs,
rtpproxy should start relaying immediately to the port specified in the
SDP. If the UA sends UDP packets from another source:IP port, it should
update internal session data, so that all subsequent packets will be
relayed to the proper IP:port. Otherwise it will continue to user the
Hi Ssergey,
the ACK seem OK, but the R-URI does not contain the "sip.instance"
parameter defined in the 200 OK's Contact. Usually all contact parameter
should be preserved when constructing the ACK R-URI.
Does your server receive the ACK, but it is not able to forward it back
to the callee (if
Hi,
have you tired SIP Session Timers
(https://kamailio.org/docs/modules/5.1.x/modules/sst.html)?
Daniel
On 08/29/2018 11:30 AM, Hamid Hashmi wrote:
Dear Users
Please advise how to detect a network lost at caller or callee side. I
have tried timeout_socket
Hi Enrico,
Are you calling it in conjunction with rtpengine? maybe this can help
you: https://lists.kamailio.org/pipermail/sr-users/2015-August/089434.html
Daniel
On 07/05/2018 02:02 PM, Enrico Bandiera wrote:
Hi, I'm encountering an issue where calling keep_codecs_by_name does
not modify