[OpenSIPS-Users] Disable an rtpengine with trafic issue

Alain Bieuzent alain.bieuzent at free.fr
Thu Sep 3 12:32:41 EST 2020


Thanks Karsten for your reply,

 

I just looked at the options for rtpengine-ctl I don't see a solution allowing the maintenance of a node.

Can you point me in the correct direction ?

 

Thanks

 

De : Users <users-bounces at lists.opensips.org> au nom de Karsten Horsmann <khorsmann at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mercredi 2 septembre 2020 à 23:44
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Disable an rtpengine with trafic issue

 

Hi Alain,

 

IMHO it should be possible on the rtpengine side to set one instance with the rtpengine-ctl perl script into maintenance mode you need. 

 

 

 

Cheers 

Karsten 

 

Alain Bieuzent <alain.bieuzent at free.fr> schrieb am Mo., 31. Aug. 2020, 15:45:

Hi,

 

I’m using opensips V3.0.3 with a pool of two rtpengine.

For maintenance reason, i need sometimes to stop an rtpengine server.

 

When I run the command to disable one of my rtpengine “opensips-cli -x mi rtpengine_enable udp: 10.207.201.25:2223 0”, if there is current traffic on it, the new delete commands sent by opensips are sent to the other rtpengine (10.207.201.24) ex :

 

10.207.201.39:49799 -> 10.207.201.25:2223

  18035_4 d3:sdp259:v=0..o=root 1293627189 1293627189 IN IP4 185.101.180.169..s=Maniterm Media Server..c=IN IP4 185.101.180.169..t=0 0..m=audio 12792 RTP/AVP 8 101..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime

  :20..a=maxptime:150..a=sendrecv..3:ICE6:remove7:replacel18:session-connection6:origine18:transport-protocol7:RTP/AVP7:call-id53:07a145324315511e2e91b80c085bf23e at 185.101.180.169:506013:received-froml3:IP415:185.101.180.169e8:from-tag10:as5d6bc41

  17:command5:offer                                                                                                                                                                                                                                   

#

U 10.207.201.25:2223 -> 10.207.201.39:49799

  18035_4 d3:sdp271:v=0..o=root 1293627189 1293627189 IN IP4 185.101.180.91..s=Maniterm Media Server..c=IN IP4 185.101.180.91..t=0 0..m=audio 37426 RTP/AVP 8 101..a=maxptime:150..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101

   0-16..a=sendrecv..a=rtcp:37427..a=ptime:20..6:result2:oke                                                                                                                                                                                          

#

U 10.207.201.39:49799 -> 10.207.201.25:2223

  18035_5 d3:sdp282:v=0..o=root 325023848 325023848 IN IP4 185.9.251.208..s=Asterisk PBX 11.11.0~dfsg-2+alphalink-1..c=IN IP4 185.9.251.208..t=0 0..m=audio 21778 RTP/AVP 8 101..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0

  -16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..3:ICE6:remove7:replacel18:session-connection6:origine18:transport-protocol7:RTP/AVP7:call-id53:07a145324315511e2e91b80c085bf23e at 185.101.180.169:506013:received-froml3:IP413:185.9.251.192e8

  :from-tag10:as3fdd699a6:to-tag10:as5d6bc4117:command5:offer                                                                                                                                                                                         

#

U 10.207.201.25:2223 -> 10.207.201.39:49799

  18035_5 d3:sdp298:v=0..o=root 325023848 325023848 IN IP4 185.101.180.91..s=Asterisk PBX 11.11.0~dfsg-2+alphalink-1..c=IN IP4 185.101.180.91..t=0 0..m=audio 37446 RTP/AVP 8 101..a=silenceSupp:off - - - -..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep

  hone-event/8000..a=fmtp:101 0-16..a=sendrecv..a=rtcp:37447..a=ptime:20..6:result2:oke                                                                                                                                                               

#

U 10.207.201.39:55934 -> 10.207.201.25:2223

  18038_5 d3:sdp282:v=0..o=root 325023848 325023848 IN IP4 185.9.251.208..s=Asterisk PBX 11.11.0~dfsg-2+alphalink-1..c=IN IP4 185.9.251.208..t=0 0..m=audio 21778 RTP/AVP 8 101..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0

  -16..a=silenceSupp:off - - - -..a=ptime:20..a=sendrecv..3:ICE6:remove7:replacel18:session-connection6:origine18:transport-protocol7:RTP/AVP7:call-id53:07a145324315511e2e91b80c085bf23e at 185.101.180.169:506013:received-froml3:IP413:185.9.251.192e8

  :from-tag10:as3fdd699a6:to-tag10:as5d6bc4117:command5:offer                                                                                                                                                                                         

#

U 10.207.201.25:2223 -> 10.207.201.39:55934

  18038_5 d3:sdp298:v=0..o=root 325023848 325023848 IN IP4 185.101.180.91..s=Asterisk PBX 11.11.0~dfsg-2+alphalink-1..c=IN IP4 185.101.180.91..t=0 0..m=audio 37446 RTP/AVP 8 101..a=silenceSupp:off - - - -..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telep

  hone-event/8000..a=fmtp:101 0-16..a=sendrecv..a=rtcp:37447..a=ptime:20..6:result2:oke                                                                                                                                                               

 

 

running command opensips-cli -x mi rtpengine_enable udp: 10.207.201.25:2223

 

 

#

U 10.207.201.39:44347 -> 10.207.201.24:2223

  18037_7 d7:call-id53:07a145324315511e2e91b80c085bf23e at 185.101.180.169:506013:received-froml3:IP413:185.9.251.192e8:from-tag10:as3fdd699a7:command6:delete                                                                                           

#

U 10.207.201.24:2223 -> 10.207.201.39:44347

  18037_7 d7:warning38:Call-ID not found or tags didn't match6:result2:oke                      

 

                                            

In that case there ghost call on stopped rtpengine, because it never received the delete message.                                                           

Is there a way to run a graceful disable (continue to send delete or new offer for current traffic, but not accepting new call)  ?

 

thanks                                 

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