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2007-10-08Only update codec information if the channel has a technology private structure.file1-2/+2
(issue #10915) Reported by: ramonpeek git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85057 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-08Update codec information as well as address when doing hold reinvites.file1-0/+5
(issue #10868) Reported by: mavince git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@85023 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-05Update the remembered RTP peer information when putting an endpoint on hold ↵file1-0/+5
or taking it off hold so that the RTP stack does not initiate a needless reinvite. (closes issue #10868) Reported by: mavince git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@84818 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-03When an RFC 2833 event is sent that we don't recognize, ignore it, don't ↵tilghman1-0/+4
queue a NULL digit (closes issue #10877) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@84581 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-21gcc 4.2 has a new set of warnings dealing with cosnt pointers. This set ofrussell1-9/+9
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2. (closes issue #10774, patch from qwell) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@83432 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-27(closes issue #10562)file1-1/+1
Reported by: idkpmiller Correct jitter value output in the CLI to be as expected. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80974 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-22(closes issue #10526)file1-1/+1
Reported by: sinistermidget Revert commit from issue #10355 and return timestamp skew to 640. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@80255 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-15(closes issue #10440)file1-1/+1
Reported by: irroot (closes issue #10454) Reported by: flo_turc Increase maximum timestamp skew to 120. 20 was apparently far too low. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@79553 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-06(closes issue #10355)file1-1/+1
Reported by: wdecarne Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@78172 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-25set the sequence number in a frame for all frame typesrizzo1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@77022 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17cast arguments to ast_log so that it builds without warnings for merussell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@75447 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17Ensure that the pointer to STUN data does not go to unaccessible memory. ↵file1-2/+2
(ASA-2007-017) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@75439 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27Only output debug information related to RTCP timestamps when RTCP debugrussell1-1/+1
is turned on (issue #10066, patch by me) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@72112 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-26Don't dereference a pointer that may be NULL here.qwell1-0/+3
Issue 10017. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@71915 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-21Do not Packet2Packet bridge if packetization settings do not allow it. ↵file1-0/+13
(issue #9117 reported by phsultan) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@70727 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-20Put the speex packetization values back in but disable it when setting up ↵file1-1/+1
the smoother. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@70360 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-19Merged revisions 69992 via svnmerge from file1-12/+17
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r69992 | file | 2007-06-19 13:00:58 -0400 (Tue, 19 Jun 2007) | 2 lines Handle the CC field in the RTP header. (issue #9384 reported by DoodleHu) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@70003 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-12Merged revisions 68921 via svnmerge from file1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r68921 | file | 2007-06-12 10:18:57 -0400 (Tue, 12 Jun 2007) | 2 lines Bring RTP back to Asterisk at the end of a native bridge no matter what. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@68922 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06Merged revisions 67649 via svnmerge from file1-1/+6
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r67649 | file | 2007-06-06 09:28:34 -0400 (Wed, 06 Jun 2007) | 2 lines Reinvite the RTP back to the Asterisk machine when the timeout happens. (issue #9888 reported by gasparz) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@67650 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-04Add a missing \n. (pointed out by jcmoore on IRC)russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@67071 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-29Handle cases where a frame may have no data. (issue #9519 reported by dmb)file1-2/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66437 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Make 1.4 build on my machine, too..russell1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@66068 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24don't use uninitialized variableskpfleming1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65965 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24I like it when the RTP stack compiles myself...file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65863 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24Fix the calculation of the RTT for RTCP. The previous code would result inrussell1-19/+37
oscillating and incorrect data. Additionally, the RTT would sometimes report negative values due to incorrect calculations. (issue #9601, patch from davetroy) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@65842 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Avoid invalid seqno cycling detection.qwell1-1/+1
Per comment from Dave Troy: This adds back in some simple typecasting I had in an earlier version which I realize now may be breaking things. Issue #9554. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61707 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-20Remove a stray debug message introduced by a recent commit.russell1-3/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61697 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-18Clean upp formatting, add some doxygen stuff while we're in cleaning mode... ↵oej1-18/+18
Thanks Kevin! git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61676 f38db490-d61c-443f-a65b-d21fe96a405b
2007-04-18Issue #9554 - Improve RTCP (Dave Troy)oej1-7/+5
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@61674 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-29Merged revisions 59357 via svnmerge from russell1-4/+8
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r59357 | russell | 2007-03-29 12:14:33 -0500 (Thu, 29 Mar 2007) | 5 lines If an error occurs when reading from an RTP socket, and the error code does not indicate that we should try again, then return NULL instead of a "null frame". This will prevent Asterisk from trying over and over again, and eventually causing the system to crash. (issue #8285, john) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59358 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-26The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in somerussell1-2/+14
because they get set in sip_hangup. So, there are common situations where the variables will not be available in the dialplan at all. So, this patch provides an alternate method for getting to this information by introducing AUDIORTPQOS and VIDEORTPQOS dialplan functions. (issue #9370, patch by Corydon76, with some testing by blitzrage) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@59207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-12Allow RFC2833 compensation to compensate for even stupider implementations ↵file1-20/+22
by queueing up the end frame at the start, not the actual end. (issue #8963 reported by AndrewZ) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58783 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-08Make early SDP seeding even smarter! We have to check codecs in the ↵file1-6/+12
make_compatible function too. (issue #9221 reported by marcelbarbulescu) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58436 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-07Ensure we have (or should have) at least one matching codec before ↵file1-1/+12
attempting early bridge SDP seeding. (issue #9221 reported by marcelbarbulescu) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@58240 f38db490-d61c-443f-a65b-d21fe96a405b
2007-03-05Preserve marker bit when P2P bridging. (issue #9198 reported by edgreenberg)file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@57768 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-07We can not reliably do P2P bridging with DTMF passing back with compensation ↵file1-0/+8
if we need to listen for DTMF frames. (issue #8962 reported by caio1982) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53434 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-07When parsing the NTP timestamp in a sender report message, you are supposed torussell1-1/+1
take the low 16 bits of the integer part, and the high 16 bits of the fractional part. However, the code here was erroneously taking the low 16 bits of the fractional part. It then shifted the result 16 bits down, so the result was always zero. This fix makes it grab the appropriate high 16 bits, instead. (issue #8991, pointed out by andre_abrantes) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53429 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-02Correct a copy/pasted error message line for RTCP.file1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53120 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01When going on hold have the side that was put on hold reinvite back to ↵file1-1/+14
Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53052 f38db490-d61c-443f-a65b-d21fe96a405b
2007-02-01Add more frame types to forward in the RTP bridge loops.file1-5/+15
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53050 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-31Merged revisions 53039 via svnmerge from russell1-9/+9
https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines Use the proper format string to print unsigned values in the rtp debug output. (issue #8954, wmis) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@53040 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29Fix a problem with packet-to-packet bridging and DTMF mode translation. P2Prussell1-1/+9
bridging can only be used when the DTMF modes don't match if the core is monitoring DTMF in both directions. Then, the core will handle the translation. Otherwise, this bridging method can not be used. (issue #8936) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@52645 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-19Merge the changes from the /team/group/vldtmf_fixup branch.russell1-10/+26
The main bug being addressed here is a problem introduced when two SIP channels using SIP INFO dtmf have their media directly bridged. So, when a DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk would try to emulate a digit of some length by first sending a DTMF BEGIN frame and sending a DTMF END later timed off of incoming audio. However, since there was no audio coming in, the DTMF_END was never generated. This caused DTMF based features to no longer work. To fix this, the core now knows when a channel doesn't care about DTMF BEGIN frames (such as a SIP channel sending INFO dtmf). If this is the case, then Asterisk will not emulate a digit of some length, and will instead just pass through the single DTMF END event. Channel drivers also now get passed the length of the digit to their digit_end callback. This improves SIP INFO support even further by enabling us to put the real digit duration in the INFO message instead of a hard coded 250ms. Also, for an incoming INFO message, the duration is read from the frame and passed into the core instead of just getting ignored. (issue #8597, maybe others...) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51311 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-18Pass data as well for hold/unhold/vidupdate frames. (issue #8840 reported by ↵file1-2/+2
mdu113) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51211 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-17Return the correct result when directly writing out a packet so that the ↵file1-2/+2
core doesn't then decide to handle it the regular way again. (issue #8833 reported by rcourtna) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51182 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-17Fix issue with dtmf continuation packets when the dtmf digit is 0...qwell1-2/+5
Issue 8831 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@51170 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-11Add support to see whether NAT was detected (yay symmetric RTP) and also add ↵file1-0/+5
a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@50466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-08Disable the more intense packet2packet bridging until the bugs can be worked ↵file1-0/+10
out. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@50032 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-30If the Packet2Packet bridge is being broken because of a masquerade then ↵file1-0/+4
attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49066 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-27since these variables all have static duration, none of them need ↵kpfleming1-7/+7
initializers (they default to zero anyway) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@49006 f38db490-d61c-443f-a65b-d21fe96a405b
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