https://origsvn.digium.com/svn/asterisk/trunk
........
r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
Only change the RTP ssrc when we see that it has changed
This change basically reverts the change reviewed in
https://reviewboard.asterisk.org/r/374/ and instead limits the
updating of the RTP synchronization source to only those times when we
detect that the other side of the conversation has changed the ssrc.
The problem is that SRCUPDATE control frames are sent many times where
we don't want a new ssrc, including whenever Asterisk has to send DTMF
in a normal bridge. This is also not the first time that this mistake
has been made. The initial implementation of the ast_rtp_new_source
function also changed the ssrc--and then it was removed because of
this same issue. Then, we put it back in again to fix a different
issue. This patch attempts to only change the ssrc when we see that
the other side of the conversation has changed the ssrc.
It also renames some functions to make their purpose more clear.
Review: https://reviewboard.asterisk.org/r/540/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@252135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r224671 | kpfleming | 2009-10-19 18:47:39 -0500 (Mon, 19 Oct 2009) | 14 lines
Merged revisions 224670 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r224670 | kpfleming | 2009-10-19 18:44:07 -0500 (Mon, 19 Oct 2009) | 7 lines
Correct timestamp calculations when RTP sample rates over 8kHz are used.
While testing some endpoints that support 16kHz and 32kHz sample rates, some
log messages were generated due to calc_rxstamp() computing timestamps in a way
that produced odd results, so this patch sanitizes the result of the
computations.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@224673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r221266 | twilson | 2009-09-30 12:52:30 -0500 (Wed, 30 Sep 2009) | 32 lines
Merged revisions 221086 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
Change the SSRC by default when our media stream changes
Be default, change SSRC when doing an audio stream changes Asterisk doesn't
honor marker bit when reinvited to already-bridged RTP streams,resulting in
far-end stack discarding packets with "old" timestamps that areactually part of
a new stream. This patch sends AST_CONTROL_SRCUPDATE whenever there is a
reinvite, unless the 'constantssrc' is set to true in sip.conf.
The original issue reported to Digium support detailed the following situation:
ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
fromITSP, Asterisk dials the app server which sends a re-invite back
toAsterisk--not to negotiate to send media directly to the ITSP, but to
indicatethat it's changing the stream it's sending to Asterisk. The app
servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
bit on the new stream. Asterisk passes through the teimstamp of the new stream,
butdoes not reset the SSRC, sequence numbers, or set the marker bit.
When the timestamp on the new stream is older than the timestamp on the
originalstream, the ITSP (which doesn't know there has been any change) discards
the newframes because it thinks they are too old. This patch addresses this by
changing the SSRC on a stream update unless constantssrc=true is set in
sip.conf.
Review: https://reviewboard.asterisk.org/r/374/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@221302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since 1.6.X still has the deprecated 'rtp debug ip <foo>'
this patch is different from the fix that went into trunk
(closes issue 0015711)
Reported by: davidw
Patches:
2009082800-rtpdebug.diff.txt uploaded by mvanbaak (license 7)
Tested by: davidw
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@218112 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@208484 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r197606 | mmichelson | 2009-05-28 10:32:19 -0500 (Thu, 28 May 2009) | 22 lines
Recorded merge of revisions 197588 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r197588 | mmichelson | 2009-05-28 10:27:49 -0500 (Thu, 28 May 2009) | 16 lines
Allow for media to arrive from an alternate source when responding to a reinvite with 491.
When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.
As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.
Review: https://reviewboard.asterisk.org/r/252
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@197618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines
Merged revisions 180372 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Review: http://reviewboard.digium.com/r/184/
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@180378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines
Merged revisions 178141 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
(closes issue #14460)
Reported by: moliveras
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@178172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines
Merged revisions 175124 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines
Don't send DTMF for infinite time if we do not receive an END event.
I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf,
dtmftimeout, that was intended to handle this situation. However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.
The default timeout is 3 seconds. However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:
Limiting the time period of extending the tone is necessary
to avoid that a tone "gets stuck". Regardless of the
algorithm used, the tone SHOULD NOT be extended by more than
three packet interarrival times. A slight extension of tone
durations and shortening of pauses is generally harmless.
Three seconds will pretty much _always_ be far more than three packet
interarrival times. However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.
Code from svn/asterisk/team/russell/issue_14460
(closes issue #14460)
Reported by: moliveras
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@175129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r162656 | file | 2008-12-10 12:06:59 -0400 (Wed, 10 Dec 2008) | 13 lines
Merged revisions 162653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162653 | file | 2008-12-10 12:05:29 -0400 (Wed, 10 Dec 2008) | 6 lines
Increment the sequence number on the end packets for RFC2833. After reading the RFC some more and doing some testing I agree with this change.
(closes issue #12983)
Reported by: vt
Patches:
dtmf_inc_seqnum_on_end_pkts.diff uploaded by vt (license 520)
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines
Merged revisions 162204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162207 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r161014 | jpeeler | 2008-12-04 12:32:20 -0600 (Thu, 04 Dec 2008) | 17 lines
Merged revisions 161013 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r161013 | jpeeler | 2008-12-04 12:30:41 -0600 (Thu, 04 Dec 2008) | 9 lines
(closes issue #13835)
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@161016 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
........
r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines
(closes issue #13557)
Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@147811 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/trunk
................
r143340 | mmichelson | 2008-09-17 13:26:35 -0500 (Wed, 17 Sep 2008) | 14 lines
Merged revisions 143337 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r143337 | mmichelson | 2008-09-17 13:24:15 -0500 (Wed, 17 Sep 2008) | 6 lines
Allow for "G.729" if offered in an SDP even though
it is not RFC 3551 compliant. Some Cisco switches
will send this in an SDP, and it doesn't hurt to
be able to accept this.
........
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@143349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines
Since adding the AST_CONTROL_SRCUPDATE frame type,
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines
Fix a problem where inbound rfc2833 audio would be sent to the
core instead of being P2P bridged. When the core regenerated
the rfc2833 packet for the outbound leg, the SSRC would be different
than the RTP audio on the call leg causing DTMF detection issues on
the far end.
(closes issue #12955)
Reported by: tonyredstone
Patches:
dynamic_rtp.patch uploaded by tsearle (license 373)
Tested by: tonyredstone
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129437 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and not use the p2p rtp bridge). I could not find a way to detect us using the p2p bridge, which
would be nice.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128197 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines
Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames.
(closes issue #11429)
Reported by: sperreault
Patches:
11429-frametype.diff uploaded by qwell (license 4)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@110020 65c4cc65-6c06-0410-ace0-fbb531ad65f3
