[OpenSIPS-Devel] [ opensips-Bugs-3584714 ] RTP Traffic Sent to wrong destination

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Tue Nov 6 17:08:29 CET 2012


Bugs item #3584714, was opened at 2012-11-06 03:13
Message generated for change (Comment added) made by bogdan_iancu
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Category: core
Group: 1.8.x
>Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: apsaras (apsaras)
Assigned to: Nobody/Anonymous (nobody)
Summary: RTP Traffic Sent to wrong destination

Initial Comment:
During the negotiation with a carrier SBC where SIP and RTP are running on different IPs, I found out that in some cases - unknown when - the RTP traffic is sent to SIP IP and not to RTP IP. We run 4 RTP Proxies one for each core in order to maximize performance but even if we use just one of them, the problem remains.

That is the initial INVITE sent from our Proxy to the Carrier

INVITE sip:4303030211176XXXX at 194.XXX.XXX.66 SIP/2.0
Record-Route: <sip:91.XXX.XXX.90;lr;nat=yes;did=716.b4738932>
Via: SIP/2.0/UDP 91.XXX.XXX.90;branch=z9hG4bK7e46.ec664a53.0
Max-Forwards: 69
From: "asterisk" <sip:asterisk at 91.XXX.XXX.90>;tag=as0b2df024
To: <sip:30211176XXXX at 91.XXX.XXX.90>
Contact: <sip:91.XXX.XXX.90;did=716.b4738932>
Call-ID: 36cbb21129aa50d82ebaef9764ca8c23 at 192.168.1.15
CSeq: 102 INVITE
User-Agent: OpenSIPs
Privacy: id;header;user
P-Asserted-Identity:  <sip:0 at 91.XXX.XXX.90>
Date: Tue, 06 Nov 2012 10:52:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 2009131870 2009131870 IN IP4 91.XXX.XXX.90
s=OpenSIPs
c=IN IP4 91.XXX.XXX.90
t=0 0
m=audio 14034 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=direction:active
a=sdpmangled:yes

and this is the OK sent from Carrier

SIP/2.0 200 OK
Session-Expires: 3600;refresher=uas
Require: timer
Via: SIP/2.0/UDP 91.XXX.XXX.90;branch=z9hG4bK7e46.ec664a53.0
Record-Route: <sip:91.XXX.XXX.90;lr;nat=yes;did=716.b4738932>
To: <sip:30211176XXXX at 91.XXX.XXX.90>;tag=3561187924-714596
From: "asterisk" <sip:asterisk at 91.XXX.XXX.90>;tag=as0b2df024
Call-ID: 36cbb21129aa50d82ebaef9764ca8c23 at 192.168.1.15
CSeq: 102 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Contact: <sip:4303030211176XXXX at 194.XXX.XXX.66:5060>
Call-Info: <sip:194.XXX.XXX.66>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 333

v=0
o=root 188 1 IN IP4 194.XXX.XXX.66
s=sip call
c=IN IP4 194.XXX.XXX.76
t=0 0
m=audio 44924 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=X-vrzcap:vbd Ver=0 Mode=FaxPr ModemRtpRed=0
a=X-vrzcap:identification bin=DSR1e814 Prot=mgcp App=MG

But RTP traffic sent to .66 instead of .76

----------------------------------------------------------------------

>Comment By: Bogdan-Andrei Iancu (bogdan_iancu)
Date: 2012-11-06 08:08

Message:
Hi,

Hard to tell without knowing:
   what flags do you pass to rtpproxy_xxxx() 
   a full capture of RTP received on that call - rtpproxy learns
destination based on the incoming traffic.

Regards,
Bogdan

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Comment By: apsaras (apsaras)
Date: 2012-11-06 08:02

Message:
Problem solved.

Need to put r flag to engage_rtp_proxy command.

----------------------------------------------------------------------

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