[SR-Users] Video calls with delay on one side

Yuriy Gorlichenko ovoshlook at gmail.com
Fri Sep 9 09:12:45 CEST 2016


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 at 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 at 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 at 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 at 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 at 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 at 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 at my.ser.ver.ip:5999]:
--- Tag 'fgl9f33gf0', created 0:39 ago for branch '', in dialogue with
'as5f26dbd9'
[1473403379.000108] INFO:
[7742bfcf0e9570ab6d7d40a52cd4cd4a at 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 at 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 at 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 at 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 at 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 at 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!
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