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: mailto:gerry.kernan@infinityit.ie gerry.kernan@infinityit.ie
Managed IT Services Infinity IT - http://www.infinityit.ie/ www.infinityit.ie
IP Telephony Asterisk Consulting - http://www.asteriskconsulting.com www.asteriskconsulting.com
Contact Centre Total Interact - http://www.totalinteract.com www.totalinteract.com
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*
cid:image001.jpg@01D105A5.2701B0E0
*Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland*
*Tel: +353 - (0)1 - 293 0090 | E-Mail: *gerry.kernan@infinityit.ie mailto:gerry.kernan@infinityit.ie**
*Managed IT Services_ _Infinity IT*- www.infinityit.ie http://www.infinityit.ie/
*IP Telephony_ _Asterisk Consulting*– www.asteriskconsulting.com http://www.asteriskconsulting.com
*Contact Centre_ _Total Interact*– www.totalinteract.com http://www.totalinteract.com
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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:1001_1@192.168.200.114:5064 |
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:1001_1@192.168.200.114:5064 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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064 Contact: sip:012930090@192.10.10.216:5060 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 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:012930090@192.10.10.216;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:1001_1@192.168.200.114:5064 |
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:1001_1@192.168.200.114:5064 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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064 Contact: sip:012930090@192.10.10.216:5060 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 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:012930090@192.10.10.216;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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064;tag=1593523975 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 CSeq: 102 INVITE Contact: sip:1001_1@192.168.200.114:5064 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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064;tag=1593523975 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 CSeq: 102 INVITE Contact: sip:1001_1@X.X.X.X: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:sr-users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin Mierla Sent: 19 April 2017 10:28 To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org 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: gerry.kernan@infinityit.ie
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 sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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:1001_1@192.168.200.114:5064 |
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:1001_1@192.168.200.114:5064 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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064 Contact: sip:012930090@192.10.10.216:5060 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 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:012930090@192.10.10.216;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:1001_1@192.168.200.114:5064 |
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:1001_1@192.168.200.114:5064 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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064 Contact: sip:012930090@192.10.10.216:5060 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 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:012930090@192.10.10.216;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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064;tag=1593523975 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 CSeq: 102 INVITE Contact: sip:1001_1@192.168.200.114:5064 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:012930090@192.10.10.216;tag=as696ac198 To: sip:1001_1@192.168.200.114:5064;tag=1593523975 Call-ID: 6024dc75117969e6677d93e44e689667@192.10.10.216:5060 CSeq: 102 INVITE Contact: sip:1001_1@X.X.X.X: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:sr-users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin Mierla Sent: 19 April 2017 10:28 To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org 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: gerry.kernan@infinityit.ie
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 sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi Daniel
Thanks for your help on this , changing direction attr on rtpengine_offer fixed my issues
Best Regards
Gerry Kernan
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Daniel-Constantin Mierla Sent: 19 April 2017 10:28 To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org 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: gerry.kernan@infinityit.ie mailto:gerry.kernan@infinityit.ie
Managed IT Services Infinity IT - http://www.infinityit.ie/ www.infinityit.ie
IP Telephony Asterisk Consulting – http://www.asteriskconsulting.com www.asteriskconsulting.com
Contact Centre Total Interact – http://www.totalinteract.com www.totalinteract.com
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gerry kernan