[asterisk-bugs] [JIRA] (ASTERISK-27843) res_pjsip_session.c: Wrong RTP port used on answer when 180 Session Progress specifies a different port

[Ross Beer (JIRA)]

Ross Beer created ASTERISK-27843:
------------------------------------

             Summary: res_pjsip_session.c: Wrong RTP port used on answer when 180 Session Progress specifies a different port
                 Key: ASTERISK-27843
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27843
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip_session
    Affects Versions: 13.20.0, GIT
         Environment: CentOS 7, Fedora 23
            Reporter: Ross Beer


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}



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