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

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24334?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24334:
------------------------------------

    Description: 
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,sip.txt:  from the Asterisk box.

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

h2. The dialplan:

{noformat}
[default]

;for testing ASTERISK-24205
exten => 354,1,Dial(SIP/354)

exten => 6001,1,Dial(SIP/6001)
{noformat}






  was:
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.

h2. The dialplan:

{noformat}
[default]

;for testing ASTERISK-24205
exten => 354,1,Dial(SIP/354)

exten => 6001,1,Dial(SIP/6001)
{noformat}







> 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: SVN, 12.5.0
>         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, pcap_chrome.pcap, sip.txt
>
>
> 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,sip.txt:  from the Asterisk box.
> pcap_chrome: pcap from the box where Chrome/SIPML5 was running.
> h2. The dialplan:
> {noformat}
> [default]
> ;for testing ASTERISK-24205
> exten => 354,1,Dial(SIP/354)
> exten => 6001,1,Dial(SIP/6001)
> {noformat}



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