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Summary:ASTERISK-10877: `sip show channels' does not display properly the codec in use for audio and video calls
Reporter:Ovidiu Sas (ovi)Labels:
Date Opened:2007-11-24 23:38:41.000-0600Date Closed:2007-12-11 09:47:23.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/CodecHandling
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:If a call is made between two registered SIP users (one with video capabilities and one without), the sip show channels displays "unkn" as a codec/Format for the subscriber with video capabilities.  This seems to the result of a double codec negotiation (see the output of the sip show channels and sip show channel).

****** ADDITIONAL INFORMATION ******

LKG7CAC92*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message  
192.168.2.109    248         5aba572c0fe  00102/00000  unkn  No       Tx: ACK                  
192.168.2.12     phone_spa3  aa269b5a-30  00101/00102  ulaw  No       Rx: ACK                  
2 active SIP channels
LKG7CAC92*CLI> sip show channel 5aba572c0fea52a666394a1428515131@192.168.2.20
LKG7CAC92*CLI>
 * SIP Call
 Curr. trans. direction:  Outgoing
 Call-ID:                5aba572c0fea52a666394a1428515131@192.168.2.20
 Owner channel ID:       SIP/248-00181c60
 Our Codec Capability:   524302
 Non-Codec Capability (DTMF):   1
 Their Codec Capability:   12
 Joint Codec Capability:   12
 Format:                 0x80004 (ulaw|h263)
 MaxCallBR:              384 kbps
 Theoretical Address:    192.168.2.109:5060
 Received Address:       192.168.2.109:5060
 SIP Transfer mode:      open
 NAT Support:            RFC3581
 Audio IP:               192.168.2.20 (local)
 Our Tag:                as6a4158f2
 Their Tag:              261e2099
 SIP User agent:         cmd_line_app 2.2.22  (VeriCall Edge)
 Username:               248
 Peername:               248
 Original uri:           sip:248@192.168.2.109
 Need Destroy:           0
 Last Message:           Tx: ACK
 Promiscuous Redir:      No
 Route:                  sip:248@192.168.2.109
 DTMF Mode:              rfc2833
 SIP Options:            (none)


LKG7CAC92*CLI> sip show channel aa269b5a-303aae87@192.168.2.12
LKG7CAC92*CLI>
 * SIP Call
 Curr. trans. direction:  Incoming
 Call-ID:                aa269b5a-303aae87@192.168.2.12
 Owner channel ID:       SIP/phone_spa3102-0017b528
 Our Codec Capability:   524302
 Non-Codec Capability (DTMF):   1
 Their Codec Capability:   3341
 Joint Codec Capability:   12
 Format:                 0x4 (ulaw)
 MaxCallBR:              384 kbps
 Theoretical Address:    192.168.2.12:5060
 Received Address:       192.168.2.12:5060
 SIP Transfer mode:      open
 NAT Support:            RFC3581
 Audio IP:               192.168.2.20 (local)
 Our Tag:                as06d26662
 Their Tag:              22a76d3e6d9b65f3o0
 SIP User agent:         Linksys/SPA3102-5.1.7(GW)
 Username:               phone_spa3102
 Peername:               phone_spa3102
 Original uri:           sip:phone_spa3102@192.168.2.12:5060
 Caller-ID:              240
 Need Destroy:           0
 Last Message:           Rx: ACK
 Promiscuous Redir:      No
 Route:                  sip:phone_spa3102@192.168.2.12:5060
 DTMF Mode:              rfc2833
 SIP Options:            replaces replace

LKG7CAC92*CLI>
Comments:By: Digium Subversion (svnbot) 2007-11-26 08:48:21.000-0600

Repository: asterisk
Revision: 89573

U   trunk/channels/chan_sip.c

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r89573 | file | 2007-11-26 08:48:21 -0600 (Mon, 26 Nov 2007) | 4 lines

Instead of printing out one codec in sip show channels print out all of the native ones (this is for video).
(closes issue ASTERISK-10877)
Reported by: ovi

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By: Ovidiu Sas (ovi) 2007-12-07 23:03:56.000-0600

Hello,

I tested this on asterisk 1.4.15 and it is half fixed.

The output of the `sip show channel <chan>' is fixed:
LKG7CAC92*CLI> sip show channel 4a56cfc0051ccc5e7a3830222a08f47d@192.168.2.20
LKG7CAC92*CLI>
 * SIP CallI>
 Curr. trans. direction:  Outgoing
 Call-ID:                4a56cfc0051ccc5e7a3830222a08f47d@192.168.2.20
 Owner channel ID:       SIP/phone_spa3102-0018e478
 Our Codec Capability:   524302
 Non-Codec Capability (DTMF):   1
 Their Codec Capability:   4
 Joint Codec Capability:   4
 Format:                 0x80004 (ulaw|h263)
 MaxCallBR:              384 kbps
 Theoretical Address:    192.168.2.12:5060
 Received Address:       192.168.2.12:5060
 SIP Transfer mode:      open
 NAT Support:            Always
 Audio IP:               192.168.2.20 (local)
 Our Tag:                as1145bd9a
 Their Tag:              3345ab07b133d031i0
 SIP User agent:        
 Username:               phone_spa3102
 Peername:               phone_spa3102
 Original uri:           sip:phone_spa3102@192.168.2.12:5060
 Need Destroy:           0
 Last Message:           Tx: ACK
 Promiscuous Redir:      No
 Route:                  sip:phone_spa3102@192.168.2.12:5060
 DTMF Mode:              rfc2833
 SIP Options:            (none)


But the output of `sip show channels' is not:
LKG7CAC92*CLI> sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message                  
192.168.2.12     phone_spa3  4a56cfc0051  00102/00000  unkn  No       Tx: ACK                  
192.168.2.109    248         46c84df99f6  00101/00101  ulaw  No       Rx: ACK                  
2 active SIP channels


See the 'unkn' under 'Form'


Regards,
Ovidiu Sas
By: Digium Subversion (svnbot) 2007-12-10 10:10:47.000-0600

Repository: asterisk
Revision: 92200

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r92200 | file | 2007-12-10 10:10:47 -0600 (Mon, 10 Dec 2007) | 4 lines

It is possible for nativeformats to contain more then one codec, so print out multiple ones.
(closes issue ASTERISK-10877)
Reported by: ovi

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=92200
By: Digium Subversion (svnbot) 2007-12-10 10:12:11.000-0600

Repository: asterisk
Revision: 92201

_U  trunk/

------------------------------------------------------------------------
r92201 | file | 2007-12-10 10:12:10 -0600 (Mon, 10 Dec 2007) | 11 lines

Blocked revisions 92200 via svnmerge

........
r92200 | file | 2007-12-10 12:13:43 -0400 (Mon, 10 Dec 2007) | 4 lines

It is possible for nativeformats to contain more then one codec, so print out multiple ones.
(closes issue ASTERISK-10877)
Reported by: ovi

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=92201
By: Digium Subversion (svnbot) 2007-12-11 09:47:23.000-0600

Repository: asterisk
Revision: 92303

_U  team/file/bridging/
U   team/file/bridging/Makefile
U   team/file/bridging/Makefile.moddir_rules
U   team/file/bridging/apps/Makefile
U   team/file/bridging/apps/app_queue.c
U   team/file/bridging/apps/app_voicemail.c
U   team/file/bridging/build_tools/make_version
U   team/file/bridging/build_tools/make_version_h
U   team/file/bridging/cdr/Makefile
U   team/file/bridging/channels/Makefile
U   team/file/bridging/channels/chan_sip.c
U   team/file/bridging/channels/chan_zap.c
U   team/file/bridging/codecs/Makefile
U   team/file/bridging/doc/CODING-GUIDELINES
U   team/file/bridging/doc/manager_1_1.txt
U   team/file/bridging/formats/Makefile
U   team/file/bridging/funcs/Makefile
U   team/file/bridging/include/asterisk/_private.h
U   team/file/bridging/include/asterisk/adsi.h
U   team/file/bridging/include/asterisk/ael_structs.h
U   team/file/bridging/include/asterisk/aes.h
U   team/file/bridging/include/asterisk/agi.h
U   team/file/bridging/include/asterisk/alaw.h
U   team/file/bridging/include/asterisk/app.h
U   team/file/bridging/include/asterisk/ast_expr.h
U   team/file/bridging/include/asterisk/astdb.h
U   team/file/bridging/include/asterisk/astobj2.h
U   team/file/bridging/include/asterisk/callerid.h
U   team/file/bridging/include/asterisk/causes.h
U   team/file/bridging/include/asterisk/cdr.h
U   team/file/bridging/include/asterisk/devicestate.h
U   team/file/bridging/include/asterisk/doxyref.h
U   team/file/bridging/include/asterisk/dsp.h
U   team/file/bridging/include/asterisk/event.h
U   team/file/bridging/include/asterisk/extconf.h
U   team/file/bridging/include/asterisk/frame.h
U   team/file/bridging/include/asterisk/hashtab.h
U   team/file/bridging/include/asterisk/io.h
U   team/file/bridging/include/asterisk/localtime.h
U   team/file/bridging/include/asterisk/logger.h
U   team/file/bridging/include/asterisk/mod_format.h
U   team/file/bridging/main/rtp.c
U   team/file/bridging/pbx/Makefile
U   team/file/bridging/res/Makefile
U   team/file/bridging/res/res_agi.c
U   team/file/bridging/utils/Makefile
U   team/file/bridging/utils/check_expr.c
U   team/file/bridging/utils/clicompat.c

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r92303 | file | 2007-12-11 09:47:21 -0600 (Tue, 11 Dec 2007) | 193 lines

Merged revisions 92082-92084,92103-92104,92122,92140,92159-92160,92199,92201,92203,92205-92206,92243,92267,92285 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

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r92082 | rizzo | 2007-12-09 23:50:38 -0400 (Sun, 09 Dec 2007) | 23 lines

Put into Makefile.moddir_rules the common instructions used to
generate loadable and embedded module lists.

Individual Makefiles now are a lot simpler, possibly as simple as this:

   -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps
   MODULE_PREFIX=cdr_
   all: _all
   include $(ASTTOPDIR)/Makefile.moddir_rules

and also more flexible because in a single directory we can combine
various types of modules (app_, cdr_, func_, ... ) by simply
listing them in the MODULE_PREFIX variable.

The individual Makefiles can also create list of modules to be
excluded by listing them in the variablel MODULE_EXCLUDE (see an
example in channels/Makefile).

With this change it becomes trivial to integrate a directory with
locally created/modified sources into the main build.



................
r92083 | rizzo | 2007-12-10 00:18:07 -0400 (Mon, 10 Dec 2007) | 7 lines

Fix the detection of modules installed from this build.

You can now add the path of local module subdirs from the command line with
  make LOCAL_MOD_SUBDIRS= ....



................
r92084 | rizzo | 2007-12-10 00:38:49 -0400 (Mon, 10 Dec 2007) | 3 lines

add a bit of info on the build infrastructure


................
r92103 | rizzo | 2007-12-10 04:35:35 -0400 (Mon, 10 Dec 2007) | 2 lines

simplify this file

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r92104 | rizzo | 2007-12-10 04:40:59 -0400 (Mon, 10 Dec 2007) | 12 lines

remove relative paths and use ASTTOPDIR instead.

Give a default value to ASTTOPDIR if unset so we can at least
do a 'make clean' without too much trouble.

The proper fix, however, is to partition the top level
Makefile in a 'setup' and a 'main' part, in a way that the
'setup' part can be included from subdirs' Makefiles and
allow targets to be built without going through the
top level Makefile.


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r92122 | rizzo | 2007-12-10 05:00:44 -0400 (Mon, 10 Dec 2007) | 2 lines

simplify/cleanup the scripts

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r92140 | oej | 2007-12-10 09:29:57 -0400 (Mon, 10 Dec 2007) | 8 lines

Add a few extra headers in the voicemail users listing in
manager 1.1. Update documentation too.

(closes issue ASTERISK-10996)
Reported by: caio1982
Patches:
     extra_vm_manager_info1.diff uploaded by caio1982 (license 22)

................
r92159 | oej | 2007-12-10 10:10:24 -0400 (Mon, 10 Dec 2007) | 24 lines

Merged revisions 92158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r92158 | oej | 2007-12-10 15:04:44 +0100 (M?�95?Y?�94?un, 10 Dec 2007) | 16 lines

Avoid reinvite race situations with two Asterisks trying
to reinvite each other in 1.4 and trunk.

This patch implements support for the 491 error code that
Asterisk 1.4 generates on situations where we get an
incoming INVITE and already has one in progress.

Thanks to mavetju for reporting and to Raj Jain for an
excellent explanation of the problem.

Patch by myself. Tested with 8 Asterisk servers connected
to each other in a training network.

Closes issue ASTERISK-10105


........

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r92160 | oej | 2007-12-10 10:18:21 -0400 (Mon, 10 Dec 2007) | 2 lines

Removing some LOG_DEBUG items

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r92199 | file | 2007-12-10 12:07:33 -0400 (Mon, 10 Dec 2007) | 4 lines

Only send a SIGHUP if the pid is greater than -1, otherwise all PIDs greater than -1 will get the SIGHUP... and that is bad.
(closes issue ASTERISK-10962)
Reported by: alanmcmillan

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r92201 | file | 2007-12-10 12:15:06 -0400 (Mon, 10 Dec 2007) | 11 lines

Blocked revisions 92200 via svnmerge

........
r92200 | file | 2007-12-10 12:13:43 -0400 (Mon, 10 Dec 2007) | 4 lines

It is possible for nativeformats to contain more then one codec, so print out multiple ones.
(closes issue ASTERISK-10877)
Reported by: ovi

........

................
r92203 | mmichelson | 2007-12-10 12:30:46 -0400 (Mon, 10 Dec 2007) | 15 lines

Merged revisions 92202 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec 2007) | 7 lines

If there are no members in a queue, then the loop where the datastore for detecting
duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means
that when we try to free it, there's a crash. This stops that crash from occurring.

(closes issue ASTERISK-10998, reported by slavon, patched by eliel)


........

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r92205 | file | 2007-12-10 12:37:35 -0400 (Mon, 10 Dec 2007) | 14 lines

Merged revisions 92204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 lines

Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much.
(closes issue ASTERISK-10985)
Reported by: revolution
Patches:
     rtp.diff uploaded by revolution (license 346)

........

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r92206 | file | 2007-12-10 12:48:18 -0400 (Mon, 10 Dec 2007) | 4 lines

Add ast_atomic_fetchadd_int_slow to check_expr for platforms that need it.
(closes issue ASTERISK-10986)
Reported by: snuffy

................
r92243 | dbailey | 2007-12-10 16:18:25 -0400 (Mon, 10 Dec 2007) | 2 lines

Add CLI commands to dynamically set hw and sw gains

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r92267 | oej | 2007-12-11 05:26:25 -0400 (Tue, 11 Dec 2007) | 2 lines

Doxygen updates

................
r92285 | oej | 2007-12-11 10:17:29 -0400 (Tue, 11 Dec 2007) | 2 lines

A lot of doxygen updates

................

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http://svn.digium.com/view/asterisk?view=rev&revision=92303