Don't ignore incoming DTLS packets after DTLS-SRTP has completed and
make sure the DTLS send queue is empties as much as possible
Change-Id: I0760c844835640385f2322767993f7eaa082bd86
This enables switching over the endpoint of a dialogue to a new client
with potentially different capabilities
Change-Id: I973b3c5d004014cc73e6ebc3e87d03c742bc1951
(cherry picked from commit 69b7586aa4)
(cherry picked from commit 93332f0f6e)
Avoid buffer overflows.
Also make cdrbuffend -= 1; append spaces to the TRUNCATED message because
syslog trims the output to 8205 chars on a line for big log buffers.
Some WebRTC clients include a=mid attributes even when BUNDLE is not in
used. This can cause problems in an answer if the offer originally used
a=mid too, as a mismatch in mid values may upset the client.
Closes#157
The payload types are not saved and retrieved from redis. Used a hash to
store the payload types in the form (0, payload_value0), (1, payload_value1)
for every media.
use via-branches to predict different destinations for branched SDP
offers without knowing the respective to-tag ahead of time.
Squashed commit of the following:
commit 0e81dc9828
Author: Richard Fuchs <rfuchs@sipwise.com>
Date: Thu May 21 11:06:15 2015 -0400
fix segfault due to missing reverse tagging
commit cd7a26314b
Author: Richard Fuchs <rfuchs@sipwise.com>
Date: Thu May 21 08:57:42 2015 -0400
support branched offer with previously unseen to-tag
commit 77da616dd2
Author: Richard Fuchs <rfuchs@sipwise.com>
Date: Wed Apr 29 11:15:37 2015 -0400
implement via-branch handling
This became necessary because of the way Asterisk handles Sequence
numbers when changing SSRC. They continue to increment a single
sequence number even though the SSRC is different, on switching back
this causes the packet_index function to interpret this as many lost
packets. The previous function had dead-spots that would not adjust
the packet_index at all if the difference fell in these ranges. These
gaps always resulted in behavior contra what would happen in webrtc
clients.