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List: asterisk-dev
Subject: [asterisk-dev] [Code Review] 4216: res_pjsip: Direct Media calls within private network sometimes ge
From: "Kevin Harwell" <reviewboard () asterisk ! org>
Date: 2014-11-26 22:17:36
Message-ID: 20141126221736.331.18337 () sonic ! digium ! api
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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/4216/
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Review request for Asterisk Developers and Joshua Colp.
Bugs: ASTERISK-24563
https://issues.asterisk.org/jira/browse/ASTERISK-24563
Repository: Asterisk
Description
-------
When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate \
network) and were bridged sometimes Asterisk would send the ip address of the \
firewall in the sdp to one of the phones in the reinvite resulting in one way audio. \
When sending the reinvite Asterisk will retrieve the media address from the \
associated rtp instance, but if frames were being read this can be overwritten with \
another address (in this case the firewall's). This patch ensures that Asterisk uses \
the original device address when using direct media.
Diffs
-----
branches/12/res/res_pjsip_sdp_rtp.c 428631
branches/12/include/asterisk/rtp_engine.h 428631
branches/12/include/asterisk/res_pjsip_session.h 428631
branches/12/channels/chan_sip.c 428631
branches/12/channels/chan_pjsip.c 428631
branches/12/bridges/bridge_native_rtp.c 428631
Diff: https://reviewboard.asterisk.org/r/4216/diff/
Testing
-------
Used a test bed of 3 phones on a private network behind a firewall with Asterisk on \
another network. Enabled direct media on the endpoints and then had phone A call \
phone B. Noted in the logged SIP reinvites that the correct address was now being \
used and also made sure audio flowed in both directions.
Thanks,
Kevin Harwell
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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/4216/">https://reviewboard.asterisk.org/r/4216/</a>
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<div>Review request for Asterisk Developers and Joshua Colp.</div>
<div>By Kevin Harwell.</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-24563">ASTERISK-24563</a>
</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" \
style="border: 1px solid #b8b5a0"> <tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: \
-moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: \
break-word;">When endpoints with direct_media enabled, behind a firewall (Asterisk on \
a separate network) and were bridged sometimes Asterisk would send the ip address of \
the firewall in the sdp to one of the phones in the reinvite resulting in one way \
audio. When sending the reinvite Asterisk will retrieve the media address from the \
associated rtp instance, but if frames were being read this can be overwritten with \
another address (in this case the firewall's). This patch ensures that Asterisk \
uses the original device address when using direct media.</pre> </td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: \
1px solid #b8b5a0"> <tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: \
-moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: \
break-word;">Used a test bed of 3 phones on a private network behind a firewall with \
Asterisk on another network. Enabled direct media on the endpoints and then had \
phone A call phone B. Noted in the logged SIP reinvites that the correct address was \
now being used and also made sure audio flowed in both directions.</pre> </td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>branches/12/res/res_pjsip_sdp_rtp.c <span style="color: \
grey">(428631)</span></li>
<li>branches/12/include/asterisk/rtp_engine.h <span style="color: \
grey">(428631)</span></li>
<li>branches/12/include/asterisk/res_pjsip_session.h <span style="color: \
grey">(428631)</span></li>
<li>branches/12/channels/chan_sip.c <span style="color: grey">(428631)</span></li>
<li>branches/12/channels/chan_pjsip.c <span style="color: grey">(428631)</span></li>
<li>branches/12/bridges/bridge_native_rtp.c <span style="color: \
grey">(428631)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/4216/diff/" style="margin-left: \
3em;">View Diff</a></p>
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