This change fixes a bug where the code which changes the transport did not check whether
the message is going out over UDP or not before changing it. For TCP and TLS transports
we don't need to change the transport as the correct one is already chosen.
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Merged revisions 410471 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Due to how messages are handled within PJSIP it is not until a message is actually
sent that the destination is reliably known. This means that the addresses placed
within the message may not be of the interface the message is being sent out on.
This module determines what interface a message is being sent on and updates the
message to contain the correct address if applicable.
This module was tested by myself in a virtualized environment with multiple interfaces
and also by Kinsey Moore in the following configuration:
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.
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.
(closes issue ASTERISK-23020)
Reported by: xrobau
Review: https://reviewboard.asterisk.org/r/3102/
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Merged revisions 410451 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp