[asterisk-bugs] [JIRA] (ASTERISK-29110) res_pjsip_sdp_rtp: Asterisk does not increment session version information in late SDP reinvite scenario

[Sebastian Damm (JIRA)]

Sebastian Damm created ASTERISK-29110:
-----------------------------------------

             Summary: res_pjsip_sdp_rtp: Asterisk does not increment session version information in late SDP reinvite scenario
                 Key: ASTERISK-29110
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29110
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip_sdp_rtp
    Affects Versions: 17.6.0
         Environment: Debian 10
            Reporter: Sebastian Damm
            Severity: Minor


Given the following scenario:

{noformat}
A --> INVITE/SDP   -->    Asterisk    -->   INVITE/SDP   -->   B
A <-- 200 OK/SDP   <--    Asterisk   <--    200 OK/SDP   <--   B
A <-- INVITE/SDP   <--    Asterisk   <--    INVITE       <--   B
{noformat}

When the reINVITE from B comes in, Asterisk answers with a 200 OK with SDP. However, when the 200 OK SDP differs from the originally sent out SDP in the INVITE, Asterisk MUST increment the session version (see https://tools.ietf.org/html/rfc4566#section-5.2), but fails to do so.

In my example the original INVITE looked like this:
{noformat}
<--- Transmitting SIP request (1262 bytes) to UDP:192.168.16.2:5060 --->
INVITE sip:5555555 at kamailio SIP/2.0
Via: SIP/2.0/UDP 192.168.16.4:5060;rport;branch=z9hG4bKPjdc71f84c-856a-4d6d-b968-b0a1a041f3d5
From: "Joe" <sip:2222222 at kamailio>;tag=469086de-cad1-40cc-a7e1-44beb4086bde
To: <sip:5555555 at kamailio>
Contact: <sip:2222222 at 192.168.16.4:5060>
Call-ID: 2ec2e851-68b9-4846-9e79-d9777ac5cbec
CSeq: 4818 INVITE
Route: <sip:kamailio;lr>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: replaces, norefersub
Max-Forwards: 70
User-Agent: Asterisk PBX 17.6.0
Proxy-Authorization: Digest username="friendlyuser", realm="kamailio", nonce="X3yarV98mYHQl7xXazqPkTLw199LIPrE", uri="sip:5555555 at kamailio", response="98fd9bc61ffef8c79604397286f03e5c"
Content-Type: application/sdp
Content-Length:   461

v=0
o=- 2042361625 2042361625 IN IP4 192.168.16.4
s=Asterisk
c=IN IP4 192.168.16.4
t=0 0
m=audio 16180 RTP/AVP 8 107 9 0 112 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:107 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
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=maxptime:60
a=sendrecv
{noformat}

The answer to the reINVITE looks like this:

{noformat}
<--- Transmitting SIP response (880 bytes) to UDP:192.168.16.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.16.2;rport=5060;received=192.168.16.2;branch=z9hG4bKd93f.ce9f801517563050b1555f627b07c15b.0
Via: SIP/2.0/UDP 192.168.16.3:35030
Call-ID: 2ec2e851-68b9-4846-9e79-d9777ac5cbec
From: <sip:5555555 at kamailio>;tag=19SIPpTag011
To: "Joe" <sip:2222222 at kamailio>;tag=469086de-cad1-40cc-a7e1-44beb4086bde
CSeq: 255 INVITE
Contact: <sip:2222222 at 192.168.16.4:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 17.6.0
Content-Type: application/sdp
Content-Length:   236

v=0
o=- 2042361625 2042361625 IN IP4 192.168.16.4
s=Asterisk
c=IN IP4 192.168.16.4
t=0 0
m=audio 16180 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
{noformat}

Expected behavior: The session version number must be increased when answering with different SDP.

To reproduce, I'll attach a docker setup. To use it:
* docker-compose build
* docker-compose up -d
* docker-compose exec sipp /bin/bash
* /testcase/start.sh
* Exit from container
* docker-compose logs asterisk

I'll attach an asterisk debug log as well.



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