Summary: | ASTERISK-27843: res_pjsip_session.c: Wrong RTP port used on 200 OK when 180 Session Progress specifies a different port | ||
Reporter: | Ross Beer (rossbeer) | Labels: | pjsip |
Date Opened: | 2018-05-08 11:58:31 | Date Closed: | 2020-01-14 11:13:44.000-0600 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Resources/res_pjsip_session |
Versions: | 13.20.0 GIT | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | CentOS 7, Fedora 23 | Attachments: | ( 0) Continued_RTP_Stream.jpg |
Description: | When a 180 Session Progress indicates a port and the subsequent 200 OK requests a different port destination RTP port. Asterisk continues to send RTP to the port specified in the 180 Session Progress.
180 Session Progress: {noformat} User Datagram Protocol, Src Port: 44228, Dst Port: 5060 Session Initiation Protocol (183) Status-Line: SIP/2.0 183 Session Progress Status-Code: 183 [Resent Packet: False] [Request Frame: 10895] [Response Time (ms): 152] Message Header From: << DATA REMOVED >> To: << DATA REMOVED >> Call-ID: << DATA REMOVED >> CSeq: 27600 INVITE Via: SIP/2.0/UDP << DATA REMOVED >> Contact: << DATA REMOVED >> User-Agent: << DATA REMOVED >> Content-Type: application/sdp Content-Length: 222 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): << DATA REMOVED >> 2197197964 0 IN IP4 << DATA REMOVED >> Session Name (s): SIP_CALL Connection Information (c): IN IP4 << DATA REMOVED >> Time Description, active time (t): 0 0 Media Description, name and address (m): audio 45056 RTP/AVP 8 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv {noformat} 200 OK - Answer: {noformat} User Datagram Protocol, Src Port: 44228, Dst Port: 5060 Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Status-Code: 200 [Resent Packet: False] [Request Frame: 10895] [Response Time (ms): 8946] Message Header From: << DATA REMOVED >> To: << DATA REMOVED >> Call-ID: << DATA REMOVED >> CSeq: 27600 INVITE Via: SIP/2.0/UDP << DATA REMOVED >> Contact: << DATA REMOVED >> User-Agent: << DATA REMOVED >> Allow: REGISTER,INVITE,ACK,BYE,REFER,NOTIFY,CANCEL,INFO,OPTIONS,PRACK,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 222 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): << DATA REMOVED >> 2197197964 1 IN IP4 << DATA REMOVED >> Session Name (s): SIP_CALL Connection Information (c): IN IP4 << DATA REMOVED >> Time Description, active time (t): 0 0 Media Description, name and address (m): audio 30024 RTP/AVP 8 101 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): sendrecv {noformat} | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-05-08 11:58:33.329-0500 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. 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]. By: Ross Beer (rossbeer) 2018-05-09 04:06:28.249-0500 Rolling back to an earlier release the issue isn't present. Looking at changes, maybe this commit has caused the issue: https://gerrit.asterisk.org/#/c/6251/2/res/res_pjsip_session.c By: Joshua C. Colp (jcolp) 2018-05-09 04:29:27.413-0500 SDP in a 200 would be an answer, not an offer. That code is also only executed in a particular DTMF scenario when it is changed through a dialplan function. By: Joshua C. Colp (jcolp) 2018-05-15 02:33:59.314-0500 The Asterisk side of things is also needed for this to see exactly what is going on with debug. By: Asterisk Team (asteriskteam) 2018-05-29 12:00:00.989-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines |