Hi. I usein kamailio 4.4 +rtpengine 4.5 for making videocalls though Web And have issue with it: I have one way audio and video till video not started. For example i calling form A point to B point B point accept call and have Audio and Video flow from the point A but point A have no any media flow. For now i using scheme when kamailio runs together with asterisk but also i had this issue with calls without asterisk. There is a link on issue i discribed before https://github.com/sipwise/rtpengine/issues/260 I use settings at the kamailio rtpengine_manage("force trust-address replace-origin replace-session-connection RTP/SAVPF") for making UDP call to WS And rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove DTLS=passive RTP/AVP media-address=MY_IP_ADDR"); For converting to back I use folloving scheme (Call from A to B) WSphone(A)->(1-st leg)-> kamailio+rtpeinge->(2-nd leg->)asterisk asterisk ->(3-st leg)-> kamailio+rtpeinge->(4-th leg)->WSphone(B) At the example below rtpengine (with kamailio) an asteirsk uses same server But connection made thoufh external interface. not local (it is made because in a production scheme asteirsk will be in a different server, but actuaaly it is does not have any matter, just for understanding topology) So at the logs after call i see next things First call from point A to asterisk (first leg and second leg) [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30170 <> A.po.int.ip:51918, 244 p, 22944 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30171 <> A.po.int.ip:51920 (RTCP), 4 p, 196 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30200 <> A.po.int.ip:51922, 209 p, 164499 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30201 <> A.po.int.ip:51924 (RTCP), 129 p, 4292 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --- Tag 'as76205e6a', created 0:39 ago for branch '', in dialogue with '5aalmrbaek' [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30156 <> my.ser.ver.ip:29236, 242 p, 23861 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30157 <> my.ser.ver.ip:29237 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30184 <> my.ser.ver.ip:27252, 148 p, 152972 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30185 <> my.ser.ver.ip:27253 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet call from asterisk to point B (3-d leg and 4-th leg) [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30238 <> my.ser.ver.ip:29958, 239 p, 24902 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30239 <> my.ser.ver.ip:29959 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30264 <> my.ser.ver.ip:40374, 205 p, 163663 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30265 <> my.ser.ver.ip:40375 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --- Tag 'fgl9f33gf0', created 0:39 ago for branch '', in dialogue with 'as5f26dbd9' [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30218 <> B.po.int.ip:26138, 242 p, 21481 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30219 <> B.po.int.ip:30161 (RTCP), 4 p, 188 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30246 <> B.po.int.ip:31174, 198 p, 163302 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30247 <> B.po.int.ip:59057 (RTCP), 12 p, 540 b, 0 e, 1473403349 last_packet As i see there is no any lost pakets and errors and all packets going through but at the asteirsk log i see that Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000102) Sent RTP P2P packet to my.ser.ver.ip:30156 (type 111, len 000071) Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000079) when packets sended from asterisk to rtpengine they have no response So it looks like that asterisk have no response from rtpengine when sending him any IP. Also i checked at the webphone side but it sending pakents. So at this side no trouble... Thanks for advices!