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

[Alexander Traud]

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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:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3882/
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> 
> (Updated July 31, 2014, 6:14 p.m.)
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> Review request for Asterisk Developers and Alexander Traud.
> 
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> Bugs: ASTERISK-18345
>     https://issues.asterisk.org/jira/browse/ASTERISK-18345
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> Repository: Asterisk
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> Description
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> 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.
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> Diffs
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>   trunk/channels/chan_sip.c 419821 
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> Diff: https://reviewboard.asterisk.org/r/3882/diff/
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> Testing
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> Made and received calls successfully with CSipSimple with full SIP headers over TLS, SRTP and multiple codecs enabled ensuring a large SDP payload.
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> Thanks,
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> ebroad
> 
>

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