> 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 > >
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