[asterisk-dev] [Code Review] SIP Session Timer: Call drops because misunderstanding of uac and uas refresher role

[Mark Michelson]

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https://reviewboard.asterisk.org/r/2118/#review7098
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/branches/1.8/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/2118/#comment13707>

    Feel free to fix the goofy spacing on these assignments :)



/branches/1.8/channels/sip/include/sip.h
<https://reviewboard.asterisk.org/r/2118/#comment13703>

    s/refresy/refresh/



/branches/1.8/configs/sip.conf.sample
<https://reviewboard.asterisk.org/r/2118/#comment13704>

    Two things:
    
    1) Red blob on line 550
    
    2) Why did you change the default from uas to uac?


- Mark


On Sept. 18, 2012, 6:12 p.m., Terry Wilson wrote:
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> (Updated Sept. 18, 2012, 6:12 p.m.)
> 
> 
> Review request for Asterisk Developers and Thomas Arimont.
> 
> 
> Summary
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> 
> From the issue:
> The SIP session timer mechanism contains a mandatory 'refresher' parameter (included in the Session-Expires header) which is used in the session timer offer/answer signaling within a SIP Invite dialog. It looks like asterisk is interpreting the uac resp. uas role only as the initial role of client and server (caller is uac, callee is uas). The standard rfc 4028 however assigns the client role to the ((RE)-Invite) requester, the server role to the ((RE)-Invite) responder.
> 
> This patch has Asterisk track the actual refresher as "us" or "them" as opposed to relying on just the configured "uas" or "uac" properties.
> 
> 
> This addresses bug AST-922.
>     https://issues.asterisk.org/jira/browse/AST-922
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> 
> Diffs
> -----
> 
>   /branches/1.8/channels/chan_sip.c 373118 
>   /branches/1.8/channels/sip/include/sip.h 373118 
>   /branches/1.8/configs/sip.conf.sample 373118 
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> Diff: https://reviewboard.asterisk.org/r/2118/diff
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> 
> Testing
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> Calls with clients supporting and not supporting session timers with Asterisk having session-refresher=uas or uac.
> 
> 
> Thanks,
> 
> Terry
> 
>

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