Hello, for this specific case, it looks like the IP addresses of the rtpproxy are used in the wrong order. Probably you need to swap the order in the rtpengine command parameters or in the parameters for rtpengine_manage() in kamailio.cfg.
Cheers, Daniel On 19.04.17 17:06, gerry kernan wrote: > Hi > Trace of inbound call to ext 1001_1 > > 1001_1 private IP 192.168.200.114 , public IP X.X.X.X > Kamailio private IP 192.10.10.202 > Kamialio Wan Y.Y.Y.Y > Asterisk private IP 192.10.10.216 > > No. Time Source Destination Protocol > Length Info > 89 23.737999 192.10.10.216 192.10.10.202 SIP/SDP > 1051 Request: INVITE sip:[email protected]:5064 | > > Frame 89: 1051 bytes on wire (8408 bits), 1051 bytes captured (8408 bits) on > interface 0 > Linux cooked capture > Internet Protocol Version 4, Src: 192.10.10.216, Dst: 192.10.10.202 > User Datagram Protocol, Src Port: 5060, Dst Port: 5060 > Session Initiation Protocol (INVITE) > Request-Line: INVITE sip:[email protected]:5064 SIP/2.0 > Message Header > Via: SIP/2.0/UDP 192.10.10.216:5060;branch=z9hG4bK7e5e19a6;rport > Max-Forwards: 70 > Route: <sip:192.10.10.202;lr;received=sip:X.X.X.X:16074> > From: "012930090" <sip:[email protected]>;tag=as696ac198 > To: <sip:[email protected]:5064> > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > User-Agent: itel > Date: Wed, 19 Apr 2017 14:35:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer, path > Remote-Party-ID: "012930090" > <sip:[email protected]>;party=calling;privacy=off;screen=no > Content-Type: application/sdp > Content-Length: 252 > Message Body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): root 740534550 740534550 IN IP4 > 192.10.10.216 > Session Name (s): Asterisk PBX 13.13.1 > Connection Information (c): IN IP4 192.10.10.216 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 18348 RTP/AVP 8 101 > Media Attribute (a): rtpmap:8 PCMA/8000 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-16 > Media Attribute (a): ptime:20 > Media Attribute (a): maxptime:150 > Media Attribute (a): sendrecv > > No. Time Source Destination Protocol > Length Info > 94 23.740557 Y.Y.Y.Y X.X.X.X SIP/SDP 1239 > Request: INVITE sip:[email protected]:5064 | > > Frame 94: 1239 bytes on wire (9912 bits), 1239 bytes captured (9912 bits) on > interface 0 > Linux cooked capture > Internet Protocol Version 4, Src: Y.Y.Y.Y, Dst: X.X.X.X > User Datagram Protocol, Src Port: 5060, Dst Port: 16074 > Session Initiation Protocol (INVITE) > Request-Line: INVITE sip:[email protected]:5064 SIP/2.0 > Message Header > Via: SIP/2.0/UDP > Y.Y.Y.Y;branch=z9hG4bK9e2a.c31a0712c4440a48beb9234062336cb8.0 > Via: SIP/2.0/UDP > 192.10.10.216:5060;received=192.10.10.216;branch=z9hG4bK7e5e19a6;rport=5060 > Max-Forwards: 69 > From: "012930090" <sip:[email protected]>;tag=as696ac198 > To: <sip:[email protected]:5064> > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > User-Agent: itel > Date: Wed, 19 Apr 2017 14:35:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, > INFO, PUBLISH, MESSAGE > Supported: replaces, timer, path > Remote-Party-ID: "012930090" > <sip:[email protected]>;party=calling;privacy=off;screen=no > Content-Type: application/sdp > Content-Length: 266 > Path: <sip:Y.Y.Y.Y;lr;received=sip:192.10.10.216:5060> > Path: <sip:Y.Y.Y.Y;lr;received=sip:192.10.10.216:5060> > Message Body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): root 740534550 740534550 IN IP4 > 192.10.10.216 > Session Name (s): Asterisk PBX 13.13.1 > Connection Information (c): IN IP4 192.10.10.202 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 30836 RTP/AVP 8 101 > Media Attribute (a): rtpmap:8 PCMA/8000 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): fmtp:101 0-16 > Media Attribute (a): ptime:20 > Media Attribute (a): maxptime:150 > Media Attribute (a): sendrecv > Media Attribute (a): rtcp:30837 > > No. Time Source Destination Protocol > Length Info > 114 27.567325 X.X.X.X Y.Y.Y.Y SIP/SDP 910 > Status: 200 OK | > > Frame 114: 910 bytes on wire (7280 bits), 910 bytes captured (7280 bits) on > interface 0 > Linux cooked capture > Internet Protocol Version 4, Src: X.X.X.X, Dst: Y.Y.Y.Y > User Datagram Protocol, Src Port: 16074, Dst Port: 5060 > Session Initiation Protocol (200) > Status-Line: SIP/2.0 200 OK > Message Header > Via: SIP/2.0/UDP > Y.Y.Y.Y;branch=z9hG4bK9e2a.c31a0712c4440a48beb9234062336cb8.0 > Via: SIP/2.0/UDP > 192.10.10.216:5060;received=192.10.10.216;branch=z9hG4bK7e5e19a6;rport=5060 > From: "012930090" <sip:[email protected]>;tag=as696ac198 > To: <sip:[email protected]:5064>;tag=1593523975 > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > Contact: <sip:[email protected]:5064> > Content-Type: application/sdp > Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, > REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE > User-Agent: Yealink SIP-T28P 2.72.23.3 > Content-Length: 217 > Message Body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): - 20106 20106 IN IP4 > 192.168.200.114 > Session Name (s): SDP data > Connection Information (c): IN IP4 192.168.200.114 > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 11780 RTP/AVP 8 101 > Media Attribute (a): rtpmap:8 PCMA/8000 > Media Attribute (a): sendrecv > Media Attribute (a): ptime:20 > Media Attribute (a): fmtp:101 0-15 > Media Attribute (a): rtpmap:101 telephone-event/8000 > > No. Time Source Destination Protocol > Length Info > 118 27.568042 192.10.10.202 192.10.10.216 SIP/SDP > 987 Status: 200 OK | > > Frame 118: 987 bytes on wire (7896 bits), 987 bytes captured (7896 bits) on > interface 0 > Linux cooked capture > Internet Protocol Version 4, Src: 192.10.10.202, Dst: 192.10.10.216 > User Datagram Protocol, Src Port: 5060, Dst Port: 5060 > Session Initiation Protocol (200) > Status-Line: SIP/2.0 200 OK > Message Header > Via: SIP/2.0/UDP > 192.10.10.216:5060;received=192.10.10.216;branch=z9hG4bK7e5e19a6;rport=5060 > From: "012930090" <sip:[email protected]>;tag=as696ac198 > To: <sip:[email protected]:5064>;tag=1593523975 > Call-ID: [email protected]:5060 > CSeq: 102 INVITE > Contact: <sip:[email protected]:16074> > Content-Type: application/sdp > Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, > REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE > User-Agent: Yealink SIP-T28P 2.72.23.3 > Content-Length: 381 > Message Body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): - 20106 20106 IN IP4 > 192.168.200.114 > Session Name (s): SDP data > Connection Information (c): IN IP4 Y.Y.Y.Y > Time Description, active time (t): 0 0 > Media Description, name and address (m): audio 30842 RTP/AVP 8 101 > Media Attribute (a): rtpmap:8 PCMA/8000 > Media Attribute (a): ptime:20 > Media Attribute (a): fmtp:101 0-15 > Media Attribute (a): rtpmap:101 telephone-event/8000 > Media Attribute (a): sendrecv > Media Attribute (a): rtcp:30843 > Media Attribute (a): candidate:iSclzzeROGDPhRK5 1 UDP 2130706431 > Y.Y.Y.Y 30842 typ host > Media Attribute (a): candidate:iSclzzeROGDPhRK5 2 UDP 2130706430 > Y.Y.Y.Y 30843 typ host > Best Regards > > Gerry Kernan > > From: sr-users [mailto:[email protected]] On Behalf Of > Daniel-Constantin Mierla > Sent: 19 April 2017 10:28 > To: Kamailio (SER) - Users Mailing List <[email protected]> > Subject: Re: [SR-Users] Kamailio rtpengine sdp > > Hello, > you have to instruct rtpengine to do bridging between the two network > interfaces. > Can you show the INVITE and the 200ok with all SDPs, both sides (incoming and > outgoing to/from kamailio), for the case you get audio problem? Then we can > confirm if the SDP has been updated properly for bridging. > Cheers, > Daniel > > On 18.04.17 17:23, gerry kernan wrote: > Hi > > Thanks in advance if anyone can point me in the correct direction . > I have kamailio running in front of some asterisk VM’s. SIP endpoint > register to their asterisk PBX via Kamailio dispatcher module. I’m running > rtpengine with a Wan and private interface to bridge audio stream between > these endpoints on the WAN to asterisk PBX running on LAN IP behind Kamailio. > Calls from ext to ext work fine audio both directions , calls outbound to > PSTN via SIP trunk to SIP provider via trunk on asterisk work fine audio both > directions. But incoming calls via SIP provider I only get audio on stream > from asterisk registered ext to external caller , no audio from external > caller to the asterisk ext. > I reckon I have something wrong in my Kamailio.cfg . if I register an ext > direct to asterisk I get audio both ways on incoming calls. And rtp logs from > rtpenegine show it as trying to send the rtp to the private address of the > sip endpoint rather that its WAN address. > I think my mistake in somewhere in the cfg below. > Do I need to handle invites from the backend asterisk servers different that > invites from sip endpoints? > > > > Gerry Kernan > > > > Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin > D18 E3C8 | Ireland > Tel: +353 - (0)1 - 293 0090 | E-Mail: [email protected] > > Managed IT Services Infinity IT - www.infinityit.ie > IP Telephony Asterisk Consulting – > www.asteriskconsulting.com > Contact Centre Total Interact – www.totalinteract.com > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com _______________________________________________ Kamailio (SER) - Users Mailing List [email protected] https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
