Re: [SR-Users] rtpproxy not forwarding rtp packets.

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

Re: [SR-Users] Whats is wrong in this call

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

Re: [SR-Users] How to detect and Hangup a call on network lost at Caller or Callee?

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

Re: [SR-Users] sdpops / keep_codecs_by_name not modifying sdp unless i call msg_apply_changes

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