[asterisk-bugs] [JIRA] (ASTERISK-24334) Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5)

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-24334:
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             Summary: Crash with chan_sip on SIP to SIP over WebSockets call (WebRTC, SIPML5)
                 Key: ASTERISK-24334
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24334
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_sip/General, Channels/chan_sip/SRTP, Channels/chan_sip/WebSocket
    Affects Versions: 12.5.0, SVN
         Environment: SVN-branch-12-r423172, Chrome (37.0.2062.120), SIPML5 SVN 224 live demo.
            Reporter: Rusty Newton
            Severity: Critical
         Attachments: backtrace.txt, full.txt, messages.txt, pcap_asterisk.pcap

Essentially the same configuration as ASTERISK-24205, but attempting to call from a local Digium phone configured as a SIP peer to the SIPML5 client in Chrome (which resides on another machine on local network).

h2. The crash!:

D Phone -> Asterisk -> Chrome(SIPML5)

With this call path we see a crash after the answer button is clicked on the SIPML5 client interface.

h2. Additional note:

The opposite call path fails seemingly the same as seen in ASTERISK-24205. The path is contrasted with ASTERISK-24205 where the call is directly to an Asterisk application rather than another device registered to Asterisk.

Chrome(SIPML5) -> Asterisk -> D Phone

h2. The debug:

Debug attached is for the crashing scenario only.

pcap_asterisk, full, messages, backtrace.txt:  from the Asterisk box.

pcap_chrome: pcap from the box where Chrome/SIPML5 was running.








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