[asterisk-bugs] [JIRA] (ASTERISK-26003) Unexpected call termination when using PJSIP + SRTP

[Ian Gilmour (JIRA)]

Ian Gilmour created ASTERISK-26003:
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             Summary: Unexpected call termination when using PJSIP + SRTP
                 Key: ASTERISK-26003
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26003
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip
    Affects Versions: 13.7.2
         Environment: centos 6 (64-bit)
asterisk 13.7.2
pjproject 2.4.5
libsrtp 1.5.2
            Reporter: Ian Gilmour


Hi, I'm seeing calls consistently terminating ~29-30mins into the call (say to the Asterisk echo()) when using chan_pjsip (with TLS) and SRTP (media_encryption=sdes). If I change pjsip.conf to use RTP (media_encryption=no) it all works fine.

Should I be using another version of libsrtp? Or some other combination of source code packages?




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