| Summary: | ASTERISK-29110: res_pjsip_sdp_rtp: Asterisk does not increment session version information in late SDP reinvite scenario | ||||
| Reporter: | Sebastian Damm (sdamm) | Labels: | |||
| Date Opened: | 2020-10-06 11:42:50 | Date Closed: | 2020-10-07 04:47:50 | ||
| Priority: | Minor | Regression? | No | ||
| Status: | Closed/Complete | Components: | Resources/res_pjsip_sdp_rtp | ||
| Versions: | 17.6.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Debian 10 | Attachments: | ( 0) asterisk.log ( 1) asteriskLateSdpReInvite.tgz | ||
| Description: | 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@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@kamailio>;tag=469086de-cad1-40cc-a7e1-44beb4086bde To: <sip:5555555@kamailio> Contact: <sip:2222222@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@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@kamailio>;tag=19SIPpTag011 To: "Joe" <sip:2222222@kamailio>;tag=469086de-cad1-40cc-a7e1-44beb4086bde CSeq: 255 INVITE Contact: <sip:2222222@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. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2020-10-06 11:42:51.478-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Sebastian Damm (sdamm) 2020-10-06 11:44:17.223-0500 Docker scenario to reproduce and asterisk debug log file By: Sean Bright (seanbright) 2020-10-06 13:34:13.896-0500 Related to ASTERISK-28452? By: Joshua C. Colp (jcolp) 2020-10-07 04:47:50.490-0500 After looking at this it is a duplicate of the issue that Sean mentioned. | ||||