Hi,

I've done some exploring of the source code and have a hunch as to what's 
happening.

The full trace is as follows where:
1.      Endpoints = 10.1.17.12 / 10.249.224.9
2.      SRS = 10.1.17.24
3.      OpenSIPS = 10.1.17.15

12      21:33:50.208348 10.1.17.15      5060    10.249.224.9    54204   SIP/SDP 
1125    Request: INVITE sip:[email protected]:54204;ob |
13      21:33:50.235594 10.249.224.9    54204   10.1.17.15      5060    SIP     
539     Status: 100 Trying |
14      21:33:50.235938 10.249.224.9    54204   10.1.17.15      5060    SIP     
725     Status: 180 Ringing |
15      21:33:50.236090 10.1.17.15      5060    10.1.17.12      5060    SIP     
641     Status: 180 Ringing |
20      21:33:53.262006 10.249.224.9    54204   10.1.17.15      5060    SIP/SDP 
1171    Status: 200 OK (INVITE) |
21      21:33:53.262560 10.1.17.15      5060    10.1.17.12      5060    SIP/SDP 
1073    Status: 200 OK (INVITE) |
22      21:33:53.263170 10.1.17.12      5060    10.1.17.15      5060    SIP     
532     Request: ACK sip:[email protected]:54204;ob |
24      21:33:53.263325 10.1.17.15      5060    10.1.17.24      5060    
SIP/SDP/XML     1026    Request: INVITE sip:10.1.17.24:5060 |

The errors logged:

Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:src_start_recording: could not start recording!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:tm_start_recording: cannot start recording!

It looks like either the callback tm_start_recording is being called too early 
on the 1XX packets or rtp_relay_copy_offer isn’t handling the unconfirmed 
session correctly?

And perhaps the invite to the SRS shouldn’t be going out if there’s no RTP 
stream?

I’m of course not an expert in the OpenSIPS architecture so I could be wrong. :)

I’d appreciate it if someone more knowledgeable could confirm.

Kind regards

Luis

Date: Tue, 23 Apr 2024 08:39:33 +0000
From: Luis Leal <[email protected]<mailto:[email protected]>>
To: "[email protected]<mailto:[email protected]>" 
<[email protected]<mailto:[email protected]>>
Subject: [OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue
Message-ID: 
<[email protected]<mailto:[email protected]>>
Content-Type: text/plain; charset="utf-8"

Hi there,

We're encountering a curious issue with SIPREC in upgrading from 3.2 to 3.4.4 
and I was hoping someone would be able to shed some light on it.

There are two symptoms:

  1.  Errors in the opensips log
  2.  SIPREC invite with correct SDP details (as per rtpengine log) but stream 
metadata missing from the XML metadata

The errors in the log are as follows:

Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:src_start_recording: could not start recording!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:tm_start_recording: cannot start recording!

The curious part is that the above error happens before the 200 OK is received. 
The relevant SIP trace is:

12           21:33:50.208348            10.1.17.15         5060     
10.249.224.9    54204   SIP/SDP               1125     Request: INVITE 
sip:[email protected]:54204;ob |

...Snip...

21           21:33:53.262560            10.1.17.15         5060     10.1.17.12  
       5060     SIP/SDP               1073     Status: 200 OK (INVITE) |

The SIPREC invite is still generated though but is missing stream details 
(participant details masked with +27XXXXXXXXX for privacy):

    Session Initiation Protocol (SIP as raw text)
    INVITE sip:10.1.17.24:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.17.15:5060;branch=z9hG4bK0235.aca30813.0
    To: sip:10.1.17.24:5060
    From: sip:10.1.17.24:5060;tag=c5d35275eae8a009626d3007dc8441a2-ce21
    CSeq: 2 INVITE
    Call-ID: B2B.364.22430.1713814432.535273629
    Max-Forwards: 70
    Content-Length: 1995
    User-Agent: OpenSIPS (3.4.4 (x86_64/linux))
    Require: siprec
    Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42
    Contact: sip:10.1.17.15:5060;+sip.src

    --OSS-unique-boundary-42
    Content-Type: application/sdp

    v=0
    o=- 7360776941148045834 7360776941148045834 IN IP4 10.1.17.8
    s=rtpengine-12-3-1-2-0-mr12-3-1-2-1-el9
    t=0 0
    m=audio 31432 RTP/AVP 8 101
    c=IN IP4 10.1.17.8
    a=label:0
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ssrc:1120210035 cname:060168be20ab122b
    a=sendonly
    a=rtcp:31433
    m=audio 36760 RTP/AVP 8 101
    c=IN IP4 10.1.17.8
    a=label:1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=sendonly
    a=rtcp:36761
    a=ptime:20

    --OSS-unique-boundary-42
    Content-Type: application/rs-metadata+xml
    Content-Disposition: recording-session

    <?xml version="1.0" encoding="UTF-8"?>
    <recording xmlns='urn:ietf:params:xml:ns:recording:1'>
     <datamode>complete</datamode>
     <session session_id="bjfrXRxoRe26JWSzzS2Cag==">
      <sipSessionID>2d409cb6-b066-4579-8c49-c6e6a7b9d600</sipSessionID>
     </session>
     <participant participant_id="FYGs/tYNSeym0Ty1p+NTIw==">
      <nameID aor=sip:[email protected]>
       <name>+27XXXXXXXXX</name>
      </nameID>
     </participant>
     <participant participant_id="F2NjJh6eQBmXVzUNZUDmMA==">
      <nameID aor=sip:[email protected]/>
     </participant>
     <sessionrecordingassoc session_id="bjfrXRxoRe26JWSzzS2Cag==">
      <associate-time>2024-04-22T21:33:50+0200</associate-time>
     </sessionrecordingassoc>
     <participantsessionassoc participant_id="FYGs/tYNSeym0Ty1p+NTIw==" 
session_id="bjfrXRxoRe26JWSzzS2Cag==">
      <associate-time>2024-04-22T21:33:50+0200</associate-time>
     </participantsessionassoc>
     <participantsessionassoc participant_id="F2NjJh6eQBmXVzUNZUDmMA==" 
session_id="bjfrXRxoRe26JWSzzS2Cag==">
      <associate-time>2024-04-22T21:33:50+0200</associate-time>
     </participantsessionassoc>
     <participantstreamassoc participant_id="FYGs/tYNSeym0Ty1p+NTIw==">
     </participantstreamassoc>
     <participantstreamassoc participant_id="F2NjJh6eQBmXVzUNZUDmMA==">
     </participantstreamassoc>
    </recording>
    --OSS-unique-boundary-42--

Is there a configuration item we're missing perhaps?

Kind regards

Luis Leal



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