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Update for certified/16.8-cert1-rc2
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Asterisk Development Team
85
ChangeLog
85
ChangeLog
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2020-02-19 14:24 +0000 Asterisk Development Team <asteriskteam@digium.com>
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* asterisk certified/16.8-cert1-rc2 Released.
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2020-02-13 13:39 +0000 [bb19e7feb5] George Joseph <gjoseph@digium.com>
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* res_pjsip_outbound_registration: Fix SRV failover on timeout
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In order to retry outbound registrations for some situations, we
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need access to the tdata from the original request. For instance,
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for 401/407 responses we need it to properly construct the
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subsequent request with the authentication. We also need it if
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we're iterating over a DNS SRV response record set so we can skip
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entries we've already tried.
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We've been getting the tdata from the server response rdata and
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transaction but that only works for the failures where there was
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actually a response (4XX, 5XX, etc). For timeouts there's no
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response and therefore no rdata or transaction from which to get
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the tdata. When processing a single A/AAAA record for a server,
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this wasn't an issue as we just retried that same server after the
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retry timer expired. If we got an SRV record set for the server
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though, without the state from the tdata, we just kept trying the
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first entry in the set repeatedly instead of skipping to the next
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one in the list.
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* Added a "last_tdata" member to the client state structure to keep
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track of the sent tdata.
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* Updated registration_client_send() to save the tdata it used into
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the client_state.
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* Updated sip_outbound_registration_response_cb() to use the tdata
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saved in client_state when we don't get a response from the
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server. We still use the tdata from the transaction when we DO
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get a response from the server so we can properly handle 4XX
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responses where our new request depends on it.
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General note on timeouts:
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Although res_pjsip_outbound_registration skips to the next record
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immediately when a timeout occurs during SRV set traversal, it's
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pjproject that determines how long to wait before a timeout is
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declared. As with other SIP message types, pjproject will continue
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trying the same server at an interval specified by "timer_t1" until
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"timer_b" expires. Both of those timers are set in the pjsip.conf
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"system" section.
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ASTERISK-28746
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Change-Id: I199b8274392d17661dd3ce3b4d69a3968368fa06
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2020-02-13 15:08 +0000 [43ca35b831] Kevin Harwell <kharwell@digium.com>
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* res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup
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There was a race condition between client initiated DTLS setup, and handling
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of server side ice completion that caused the underlying SSL object to get
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cleared during DTLS initialization. If this happened Asterisk would be left
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in a partial DTLS setup state. RTP packets were sent and received, but were
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not being encrypted and decrypted. This resulted in no audio, or static.
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Specifically, this occurred when '__rtp_recvfrom' was processing the handshake
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sequence from the client to the server, and then 'ast_rtp_on_ice_complete'
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gets called from another thread and clears the SSL object when calling the
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'dtls_perform_setup' function. The timing had to be just right in the sense
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that from the external SSL library perspective SSL initialization completed
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(rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks
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to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks
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it is not finished, thus not completing 'dtls_srtp_setup'.
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This patch removes calls to 'dtls_perform_setup', which clears the SSL object,
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in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear
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the underlying SSL object. If an ice candidate changes a full protocol level
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renegotiation occurs. Also, in the case of bundled ICE candidates are reused
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when a stream is added. So no real reason to have to clear, and reset in this
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instance.
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Also, this patch adds a bit of extra logging to aid in diagnosis of any future
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problems.
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ASTERISK-28742 #close
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Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f
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2020-02-07 20:42 +0000 Asterisk Development Team <asteriskteam@digium.com>
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* asterisk certified/16.8-cert1-rc1 Released.
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14
asterisk-certified-16.8-cert1-rc2-summary.html
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asterisk-certified-16.8-cert1-rc2-summary.html
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<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-certified/16.8-cert1-rc2</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-certified/16.8-cert1-rc2</h3><h3 align="center">Date: 2020-02-19</h3><h3 align="center"><asteriskteam@digium.com></h3><hr><h2 align="center">Table of Contents</h2><ol>
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<li><a href="#summary">Summary</a></li>
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<li><a href="#contributors">Contributors</a></li>
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<li><a href="#closed_issues">Closed Issues</a></li>
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<li><a href="#diffstat">Diffstat</a></li>
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</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release is a point release of an existing major version. The changes included were made to address problems that have been identified in this release series, or are minor, backwards compatible new features or improvements. Users should be able to safely upgrade to this version if this release series is already in use. Users considering upgrading from a previous version are strongly encouraged to review the UPGRADE.txt document as well as the CHANGES document for information about upgrading to this release series.</p><p>The data in this summary reflects changes that have been made since the previous release, asterisk-certified/16.8-cert1-rc1.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
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<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
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<tr valign="top"><td width="33%">1 Kevin Harwell <kharwell@digium.com><br/>1 George Joseph <gjoseph@digium.com><br/></td><td width="33%"><td width="33%">1 Kevin Harwell <kharwell@digium.com><br/>1 George Joseph <gjoseph@digium.com><br/></td></tr>
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</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Bug</h3><h4>Category: Resources/res_pjsip_outbound_registration</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28746">ASTERISK-28746</a>: res_pjsip_outbound_registration keeps retrying the first entry in a SRV record set<br/>Reported by: George Joseph<ul>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=bb19e7feb5b33c8696a3a57be900f75fe36c009f">[bb19e7feb5]</a> George Joseph -- res_pjsip_outbound_registration: Fix SRV failover on timeout</li>
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</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-28742">ASTERISK-28742</a>: res_rtp_asterisk: static for audio due to incomplete dtls/srtp setup<br/>Reported by: Kevin Harwell<ul>
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<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=43ca35b8312c9c9be740bda4b961262e751c5a54">[43ca35b831]</a> Kevin Harwell -- res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup</li>
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</ul><br><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>res_pjsip_outbound_registration.c | 49 ++++++++++++++++++++++++++++++++++++--
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1 file changed, 47 insertions(+), 2 deletions(-)</pre><br></html>
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92
asterisk-certified-16.8-cert1-rc2-summary.txt
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asterisk-certified-16.8-cert1-rc2-summary.txt
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Release Summary
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asterisk-certified/16.8-cert1-rc2
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Date: 2020-02-19
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<asteriskteam@digium.com>
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----------------------------------------------------------------------
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Table of Contents
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1. Summary
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2. Contributors
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3. Closed Issues
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4. Diffstat
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----------------------------------------------------------------------
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Summary
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[Back to Top]
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This release is a point release of an existing major version. The changes
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included were made to address problems that have been identified in this
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release series, or are minor, backwards compatible new features or
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improvements. Users should be able to safely upgrade to this version if
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this release series is already in use. Users considering upgrading from a
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previous version are strongly encouraged to review the UPGRADE.txt
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document as well as the CHANGES document for information about upgrading
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to this release series.
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The data in this summary reflects changes that have been made since the
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previous release, asterisk-certified/16.8-cert1-rc1.
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----------------------------------------------------------------------
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Contributors
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[Back to Top]
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This table lists the people who have submitted code, those that have
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tested patches, as well as those that reported issues on the issue tracker
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that were resolved in this release. For coders, the number is how many of
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their patches (of any size) were committed into this release. For testers,
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the number is the number of times their name was listed as assisting with
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testing a patch. Finally, for reporters, the number is the number of
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issues that they reported that were affected by commits that went into
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this release.
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Coders Testers Reporters
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1 Kevin Harwell 1 Kevin Harwell
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1 George Joseph 1 George Joseph
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----------------------------------------------------------------------
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Closed Issues
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[Back to Top]
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This is a list of all issues from the issue tracker that were closed by
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changes that went into this release.
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Bug
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Category: Resources/res_pjsip_outbound_registration
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ASTERISK-28746: res_pjsip_outbound_registration keeps retrying the first
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entry in a SRV record set
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Reported by: George Joseph
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* [bb19e7feb5] George Joseph -- res_pjsip_outbound_registration: Fix SRV
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failover on timeout
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Category: Resources/res_rtp_asterisk
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ASTERISK-28742: res_rtp_asterisk: static for audio due to incomplete
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dtls/srtp setup
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Reported by: Kevin Harwell
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* [43ca35b831] Kevin Harwell -- res_rtp_asterisk: bad audio (static) due
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to incomplete dtls/srtp setup
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----------------------------------------------------------------------
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Diffstat Results
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[Back to Top]
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This is a summary of the changes to the source code that went into this
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release that was generated using the diffstat utility.
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res_pjsip_outbound_registration.c | 49 ++++++++++++++++++++++++++++++++++++--
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1 file changed, 47 insertions(+), 2 deletions(-)
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