Hi Alexey,


I add the same default yesterday and solved it by deleting "codec-strip" flags.

rtpengine_manage("RTP/AVP replace-session-connection replace-origin 
codec-mask-PCMA transcode-PCMU ICE=remove");



I hope this will help you.



Regards



Le 21/06/2018 10:31, « Users au nom de Alexey Kazantsev via Users » 
<[email protected] au nom de [email protected]> a écrit :



    And there's no any errors in RTPEngine log.

    

    Jun 21 13:16:21 debian-opensips rtpengine[7291]: INFO: [860143bf5c80080e]: 
Received command 'offer' from 10.145.213.88:55706

    Jun 21 13:16:21 debian-opensips rtpengine[7291]: NOTICE: 
[860143bf5c80080e]: Creating new call

    Jun 21 13:16:21 debian-opensips rtpengine[7291]: INFO: [860143bf5c80080e]: 
Enabling transcoding engine

    Jun 21 13:16:21 debian-opensips rtpengine[7291]: INFO: [860143bf5c80080e]: 
Replying to 'offer' from 10.145.213.88:55706 (elapsed time 0.008806 sec)

    Jun 21 13:16:21 [8057] new branch at 
sip:[email protected];line=3870aa6a38323e2

    Jun 21 13:16:21 [8056] incoming reply

    Jun 21 13:16:21 [8055] incoming reply

    Jun 21 13:16:24 debian-opensips rtpengine[7291]: INFO: [861278282]: 
Received command 'offer' from 10.145.213.88:55706

    Jun 21 13:16:24 debian-opensips rtpengine[7291]: NOTICE: [861278282]: 
Creating new call

    Jun 21 13:16:24 debian-opensips rtpengine[7291]: INFO: [861278282]: 
Enabling transcoding engine

    Jun 21 13:16:24 debian-opensips rtpengine[7291]: INFO: [861278282]: 
Replying to 'offer' from 10.145.213.88:55706 (elapsed time 0.017746 sec)

    Jun 21 13:16:24 [8057] new branch at sip:[email protected]:5060;transport=udp

    Jun 21 13:16:24 [8055] incoming reply

    

    

    

    

    I also found this 
https://github.com/OpenSIPS/opensips/issues/1288#issuecomment-367293070

    

    And tried as Razvan adviced:

    

    # i know that B side talks PCMU only, so i configre it

    rtpengine_offer("RTP/AVP replace-origin replace-session-connection 
ICE=remove transcode-PCMU");

    

    

    In this case - a little bit better - the call is not dropped, and I see how 
OpenSIPS/RTPEngine

    changes the codecs, and even see RTP, but going to one side only, but in 
different codecs.

    http://rgho.st/6nkHxTW5p

    

    But still no sound in any direction  :D

    

    -----------------------------------------------

    BR, Alexey

    http://alexeyka.zantsev.com/

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