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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3102/">https://reviewboard.asterisk.org/r/3102/</a>
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<p style="margin-top: 0;">On March 10th, 2014, 3:38 p.m. CDT, <b>opticron</b> wrote:</p>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">This solved the issues I was having when using 0.0.0.0 as the bind address for the only defined transport on my multihomed setup. Previously, the issue this fixes was causing all kinds of weirdness when trying to connect calls and resulted in a hangup a few tens of seconds into the call after a delay setting up the audio path (this was a result of responses going to an incorrect address due to a contact header being wrong). After adding this module, everything cleared up.</pre>
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Some further details on the setup I used:
Networks:
* 10.24.16.0/21
** hard phone
** default gateway
* 10.24.64.0/21
** softphone with pjsip-based stack
Transport details:
bind address: 0.0.0.0
protocol: UDP
All endpoints were tested with explicitly configured transports and unconfigured transports.
Asterisk version: Trunk r410395 with and without this patch applied
This was tested with inbound and outbound calls, both of which were experiencing detrimental effects from incorrect IP addresses in SIP messages. These effects were only experienced by the soft phone on the 10.24.64.0 network since the messages to the hard phone on the 10.24.16.0 network had the correct IP address.</pre>
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<p>- opticron</p>
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<p>On February 23rd, 2014, 11:57 a.m. CST, Joshua Colp wrote:</p>
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<div>Review request for Asterisk Developers.</div>
<div>By Joshua Colp.</div>
<p style="color: grey;"><i>Updated Feb. 23, 2014, 11:57 a.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-23020">ASTERISK-23020</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<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;">This module adds messaging rewriting for multihomed situations. When messages are sent out it determines what interface and transport the message is going out on. The IP address for this is then placed into the message.
I've written this as a module since it does add a bit of additional logic for outgoing messages.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<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;">Sent and received calls/traffic in two configurations.
1. Two transports each bound to specific interfaces
2. One transport bound to 0.0.0.0
Confirmed that the correct IP address for each interface went into the message.
Some additional testing by others would be nice.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/12/res/res_pjsip_multihomed.c <span style="color: grey">(PRE-CREATION)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3102/diff/" style="margin-left: 3em;">View Diff</a></p>
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