19 Apr
2017
19 Apr
'17
6:06 p.m.
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(a)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(a)infinityit.ie
Managed IT Services Infinity IT - www.infinityit.ie
IP Telephony Asterisk Consulting – www.asteriskconsulting.com
Contact Centre Total Interact – www.totalinteract.com
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Daniel-Constantin Mierla
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