[SR-Users] Kamailio rtpengine sdp

Daniel-Constantin Mierla miconda at gmail.com
Thu Apr 20 14:05:21 CEST 2017


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 at 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 at 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 at 192.10.10.216>;tag=as696ac198
>         To: <sip:1001_1 at 192.168.200.114:5064>
>         Contact: <sip:012930090 at 192.10.10.216:5060>
>         Call-ID: 6024dc75117969e6677d93e44e689667 at 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 at 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 at 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 at 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 at 192.10.10.216>;tag=as696ac198
>         To: <sip:1001_1 at 192.168.200.114:5064>
>         Contact: <sip:012930090 at 192.10.10.216:5060>
>         Call-ID: 6024dc75117969e6677d93e44e689667 at 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 at 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 at 192.10.10.216>;tag=as696ac198
>         To: <sip:1001_1 at 192.168.200.114:5064>;tag=1593523975
>         Call-ID: 6024dc75117969e6677d93e44e689667 at 192.10.10.216:5060
>         CSeq: 102 INVITE
>         Contact: <sip:1001_1 at 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 at 192.10.10.216>;tag=as696ac198
>         To: <sip:1001_1 at 192.168.200.114:5064>;tag=1593523975
>         Call-ID: 6024dc75117969e6677d93e44e689667 at 192.10.10.216:5060
>         CSeq: 102 INVITE
>         Contact: <sip:1001_1 at 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 at lists.kamailio.org] On Behalf Of Daniel-Constantin Mierla
> Sent: 19 April 2017 10:28
> To: Kamailio (SER) - Users Mailing List <sr-users at 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
>  
>
>  
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-- 
Daniel-Constantin Mierla
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