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trunk/channels/chan_sip.c <https://reviewboard.asterisk.org/r/3882/#comment23360> 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). - Alexander Traud On July 31, 2014, 6:14 p.m., ebroad wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3882/ > ----------------------------------------------------------- > > (Updated July 31, 2014, 6:14 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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