This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.
(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.
(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
rtp-timestamp.patch uploaded by pbertera (License 5943)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When sending RTP packets via multicast the amount of data sent is stored in a variable and returned
from the write function. This is incorrect as any non-zero value returned is considered a failure while
a return value of 0 is success. For callers (such as ast_streamfile) that checked the return value
they would have considered it a failure when in reality nothing went wrong and it was actually a success.
The write function for the multicast RTP engine now returns -1 on failure and 0 on success, as it should.
(closes issue ASTERISK-17254)
Reported by: wybecom
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Sequence number was handled as an unsigned integer (usually 32 bits I think, more
depending on the architecture) and was put into the rtp packet which is basically
just a bunch of bits using an or operation. Sequence number only has 16 bits
allocated to it in an RTP packet anyway, so it would add to the next field which
just happened to be the codec. This makes sure the sequence number is set to be
a 16 bit integer regardless of architecture (hopefully) and also makes it so the
incrementing of the sequence number does bitwise or at the peak of a 16 bit number
so that the value will be set back to 0 when going beyond 65535 anyway.
(closes issue ASTERISK-18291)
Reported by: Will Schick
Review: https://reviewboard.asterisk.org/r/1542/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@342602 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a side project I've been poking at this week. The intent is to discuss
Asterisk architecture in a top down fashion to help new developers understand how
Asterisk is put together. There is a ton of stuff to write about, so this will
just continue to evolve over time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
