[asterisk-dev] [Code Review] 3882: Replace sip_tls_read() and resolve the large SDP poll issue

[ebroad]

> On Aug. 4, 2014, 1:17 p.m., Alexander Traud wrote:
> > trunk/channels/chan_sip.c, line 3041
> > <https://reviewboard.asterisk.org/r/3882/diff/1/?file=65918#file65918line3041>
> >
> >     Thank you for adding me to the list of reviewers. That way, I got E-mail notifications.
> >     No no-go from my (limited set of personal) test-cases, because they all passed successfully.
> >     
> >     Except: Case (res < 0 && errno == EAGAIN) is missing which leads to readbuf[-1] = '\0' right now. To prevent this, I went for
> >     
> >     if (res < 0) {
> >         if (errno == EAGAIN) {
> >             continue;
> >         } else {
> >             ast_debug(…
> >             return -1;
> >         }
> >     
> >     I went for "continue" because that was required to fix an issue with Nokia Symbian/S60 connected over UMTS (SIP INVITE = two TCP messages, 1.3 seconds latency between those messages); an issue introduced by the new variable "exclusive_input" (revision 416071).

Alex -
 Thank you for testing and your comments. I have implemented your change as it is more elegant, and added handling for EINTR per rmudgett's advice.


- ebroad


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3882/#review12970
-----------------------------------------------------------


On Aug. 5, 2014, 10:21 p.m., ebroad wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3882/
> -----------------------------------------------------------
> 
> (Updated Aug. 5, 2014, 10:21 p.m.)
> 
> 
> Review request for Asterisk Developers and Alexander Traud.
> 
> 
> Bugs: ASTERISK-18345
>     https://issues.asterisk.org/jira/browse/ASTERISK-18345
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> Replace sip_tls_read() and sip_tcp_read() with a single function and resolve the poll/wait issue with large SDP payloads. See https://reviewboard.asterisk.org/r/3653/ for the discussion on this.
> 
> 
> Diffs
> -----
> 
>   trunk/channels/chan_sip.c 419821 
> 
> Diff: https://reviewboard.asterisk.org/r/3882/diff/
> 
> 
> Testing
> -------
> 
> Made and received calls successfully with CSipSimple with full SIP headers over TLS, SRTP and multiple codecs enabled ensuring a large SDP payload.
> 
> 
> Thanks,
> 
> ebroad
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140806/453e784a/attachment-0001.html>