| Summary: | ASTERISK-13404: Codec negotiation fails on calls from 1.2 -> 1.6, and is sub-optimum on calls from 1.6->1.6 | ||
| Reporter: | John Covert (jcovert) | Labels: | |
| Date Opened: | 2009-01-20 01:25:32.000-0600 | Date Closed: | 2009-02-20 14:34:12.000-0600 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_iax2 |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | #1 is a major problem, although there is a work-around. #2 is minor. (1) Given a 1.2 system with the following codec settings: disallow=all ; Dis-Allow all codecs ; allow=ULAW ; except for this list of Codecs allow=ALAW allow=GSM allow=G726 allow=ADPCM allow=LPC10 allow=SPEEX allow=ILBC and a 1.6 system with allow=g722 allow=ulaw allow=alaw allow=G726 disallow=adpcm ; Horrid. --jrc disallow=lpc10 ; Icky sound quality... Mr. Roboto. allow=gsm ; Always allow GSM, it's cool :) allow=iLBC ; if it's all they've got And the default codecpriority = host setting, we see the following on a call from 1.2 -> 1.6 : -- Accepting UNAUTHENTICATED call from 12.237.35.32: > requested format = ulaw, > requested prefs = (ulaw|alaw|gsm|g726|adpcm|lpc10|speex|ilbc), > actual format = g722, > host prefs = (g722|ulaw|alaw|g726|gsm|ilbc), > priority = mine But since 1.2 can't do G.722, we end up with no audio, and a stream of the following messages: [Jan 20 00:04:25] NOTICE[13875]: channel.c:2714 __ast_read: Dropping incompatible voice frame on IAX2/12.237.35.32:4569-6213 of format ulaw since our native format has changed to 0x1000 (g722) (2) If the call comes from a device on a 1.6 system which can do ulaw but not G.722, and this system has "bandwidth=high" set to accomodate ulaw and "allow=ulaw,allow=gsm" but does not include g.722 because it has no g.722 clients, when it connects to the 1.6 system in (1) above, we see -- Hungup 'IAX2/jrclaptop-14482' -- Accepting AUTHENTICATED call from 192.168.0.4: > requested format = ulaw, > requested prefs = (ulaw|gsm), > actual format = g722, > host prefs = (g722|ulaw|alaw|g726|gsm|ilbc), > priority = mine and as a result, we do a lot of unecessary transcoding. The workaround given only these two cases is to set codecpriority to caller or to reqonly, but there are obviously other cases where these will cause problems. It seems that the default "host" priority should (as it says) give priority to the host's codec, but not if that codec is not in the requested list at all. If the host's first priority was not one of the requested codecs, but the host's second (or third) priority is one of the requested, we should take that. Maybe this needs to be a different option, but as it stands, unless the requesting host explicitly disallows the receiving host's choice (which the disallow=all in 1.2 unfortunately does not do) we end up with something not at all requested (and possibly not supported) by the originator. | ||
| Comments: | By: Digium Subversion (svnbot) 2009-02-20 14:17:40.000-0600 Repository: asterisk Revision: 177696 U branches/1.4/channels/chan_iax2.c U branches/1.4/include/asterisk/frame.h ------------------------------------------------------------------------ r177696 | dvossel | 2009-02-20 14:17:39 -0600 (Fri, 20 Feb 2009) | 8 lines Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue ASTERISK-13404) Reported by: jcovert ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=177696 By: Digium Subversion (svnbot) 2009-02-20 14:18:41.000-0600 Repository: asterisk Revision: 177697 _U trunk/ ------------------------------------------------------------------------ r177697 | dvossel | 2009-02-20 14:18:41 -0600 (Fri, 20 Feb 2009) | 13 lines Blocked revisions 177696 via svnmerge ........ r177696 | dvossel | 2009-02-20 14:17:37 -0600 (Fri, 20 Feb 2009) | 8 lines Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue ASTERISK-13404) Reported by: jcovert ........ ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=177697 By: Digium Subversion (svnbot) 2009-02-20 14:27:46.000-0600 Repository: asterisk Revision: 177698 U branches/1.6.0/channels/chan_iax2.c U branches/1.6.0/include/asterisk/frame.h ------------------------------------------------------------------------ r177698 | dvossel | 2009-02-20 14:27:46 -0600 (Fri, 20 Feb 2009) | 7 lines Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 and 15 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue ASTERISK-13404) Reported by: jcovert ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=177698 By: Digium Subversion (svnbot) 2009-02-20 14:34:12.000-0600 Repository: asterisk Revision: 177700 U branches/1.6.1/channels/chan_iax2.c U branches/1.6.1/include/asterisk/frame.h ------------------------------------------------------------------------ r177700 | dvossel | 2009-02-20 14:34:12 -0600 (Fri, 20 Feb 2009) | 8 lines Fixes issue with undefined audio codecs in chan_iax2 During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.6 only audio codec bits 0-12 are defined, leaving bits 13-14 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-14 are not defined, these bits are never turned off. In trunk, bits 13-14 are defined, which means 1.6 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities. (closes issue ASTERISK-13404) Reported by: jcovert ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=177700 | ||