https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4 lines
(closes issue #10355)
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.
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sockets other than RTP ones.
The main change is a new API function in main/rtp.c (see there
for a description)
int ast_stun_request(int s, struct sockaddr_in *dst,
const char *username, struct sockaddr_in *answer)
which can be used to send an STUN request on a socket, and
optionally wait for a reply and store the STUN_MAPPED_ADDRESS
into the 'answer' argument (obviously, the version that
waits for a reply is blocking, but this is no different
from DNS resolutions).
Internally there are minor modifications to let stun_handle_packet()
be somewhat configurable on how to parse the body of responses.
At the moment i am not committing any change to the clients,
but adding STUN client support is extremely simple, e.g. chan_sip.c
could do something like this:
+ add a variable to store the stun server address;
static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */
+ add code to parse a config file of the form "stunaddr=my.stun.server.org:3478"
(not shown for brevity);
+ right after binding the main sip socket, talk to the stun server to
determine the externally visible address
if (stunaddr.sin_addr.s_addr != 0)
ast_stun_request(sipsock, &stunaddr, NULL, &externip);
so now 'externip' is set with the externally visible address.
so it is really trivial.
Similarly ast_stun_request could be called when creating the RTP
socket (possibly adding a struct sockaddr_in field in the struct
ast_rtp to store the externalip).
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+ mark a potentially dangerous write-past-end-of-buffer
+ localize some variables in the block generating stun replies.
As before, not ready yet for a merge to 1.4
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ast_rtp_new_with_bindaddr():
1. add comments to the logic of the main loop;
2. use a common exit point on failure so the cleanup is done only in one place;
3. handle failures in rtp_socket() in the main loop of the function;
No functional changes except for #3 above, so it is not yet
worthwhile merging this and other changes to 1.4
Once the cleanup work on this file will be complete (which among
other things should include some extensions to the stun support)
it might be a good thing to push all the changes to 1.4
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(which is very little, at the moment).
Eventually, when the functionality is extended, the changes can be merged
back to 1.4. At the moment this is pointless.
Note, this change is whitespace only.
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r71063 | murf | 2007-06-22 08:10:24 -0600 (Fri, 22 Jun 2007) | 1 line
My conditions for merging amaflags info was naive; DOCUMENTATION is the default, although null is possible; theft of user-settable fields is not good. Just copy them, leave them alone.
This is for bug 10016. (plus a small fix to rtp, to elim a compiler warning (dev mode))
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Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65842 | russell | 2007-05-24 09:49:05 -0500 (Thu, 24 May 2007) | 5 lines
Fix the calculation of the RTT for RTCP. The previous code would result in
oscillating and incorrect data. Additionally, the RTT would sometimes report
negative values due to incorrect calculations.
(issue #9601, patch from davetroy)
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r61707 | qwell | 2007-04-20 16:35:27 -0500 (Fri, 20 Apr 2007) | 8 lines
Avoid invalid seqno cycling detection.
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.
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r59358 | russell | 2007-03-29 12:17:41 -0500 (Thu, 29 Mar 2007) | 13 lines
Merged revisions 59357 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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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)
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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
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)
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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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r53429 | russell | 2007-02-07 11:39:31 -0600 (Wed, 07 Feb 2007) | 7 lines
When parsing the NTP timestamp in a sender report message, you are supposed to
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)
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r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines
When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.
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r52645 | russell | 2007-01-29 15:26:27 -0600 (Mon, 29 Jan 2007) | 6 lines
Fix a problem with packet-to-packet bridging and DTMF mode translation. P2P
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)
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
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...)
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
Add support to see whether NAT was detected (yay symmetric RTP) and also add 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)
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r49066 | file | 2006-12-30 00:46:57 -0500 (Sat, 30 Dec 2006) | 2 lines
If the Packet2Packet bridge is being broken because of a masquerade then attempt to read a frame in so the masquerade actually happens. Otherwise weirdness will occur. (issue #8696 reported by kjotte)
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines
Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman)
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implementing T.140 support in RTP.
T.140/RFC 4351 is TDD over IP - text telephony for hearing impaired.
It defines a realtime text chat, much like the old "talk" application
in Unix.
T.140 is character by character in real time. It's not
the same as our current MESSAGE format - that is more like IM, but
can be gatewayed to MESSAGE with a text "codec" if needed.
More patches will follow, as soon as we've separated this code from
the video capabilities functions in the videocaps branch.
Code by John Martin, Aupix (disclaimer on file)
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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have a clear understanding of the frame allocation/deallocation, so I just mark this
for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts...
- Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting
rtcp this way, but will need feedback from rtcp gurus. This should work for
video calls too, and possibly UDPTL.
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I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too,
but feel free to backport if you see it that way. RTCP now binds to
ALL IP addresses on the host, RTP to a specific address.
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Joshua Colp
Tilghman Lesher
Russell Bryant
Luigi Rizzo
Steve Murphy
Jason Parker
Olle Johansson
Kevin P. Fleming