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
