[Dietfrid Mali <karx11erx@xxxxxxxxxxx>]

What about G.722 delivering a 16 khz audio stream, and not 8 khz as is hardcoded in rtp_player.c's device opening call to PortAudio currently?

Where is an explanation about the string tags used in register_codec_module? Can I use something like "g726_32"?

> From: tomas.kukosa@xxxxxxxxxxxxxxxxxxxxxx
> To: wireshark-dev@xxxxxxxxxxxxx
> Date: Thu, 27 Jan 2011 10:02:37 +0100
> Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
>
> Hi,
>
> I would recommend to use codec table (see epan/codecs.h) for new codecs instead of including them hardcoded into the rtp_player.
> It should not require any changes in the rtp_player for any new codec.
> Only changes for dynamic payload handling should be implemented but it is common for all codecs.
>
> The codec table is used just for codec plugins now but using for internal Wireshak codec would be good too and does not need many changes (I hope ;-).
>
> Example of G.722 implemented inside plugin (using external library) is here:
> http://anonsvn.wireshark.org/viewvc/trunk/plugins/easy_codec/easy_codec_plugin.c?view=markup
> http://anonsvn.wireshark.org/viewvc/trunk/plugins/easy_codec/codec-g722.c?view=markup
>
> Wireshak internal implementation using spandsp library could be very similar.
>
> Best regards,
> Tomas
>
>
> -----Original Message-----
> From: wireshark-dev-bounces@xxxxxxxxxxxxx [mailto:wireshark-dev-bounces@xxxxxxxxxxxxx] On Behalf Of Jaap Keuter
> Sent: Wednesday, January 26, 2011 10:22 PM
> To: Developer support list for Wireshark
> Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
>
> Hi,
>
> Found it, I'll have a look.
>
> Thanks,
> Jaap
>
> On 01/26/2011 04:59 PM, Dietfrid Mali wrote:
> > G.722 and G.726 (-32) codec integration using spandsp:
> > https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=5619
> >
> > > Date: Wed, 26 Jan 2011 00:09:50 +0100
> > > From: jaap.keuter@xxxxxxxxx
> > > To: wireshark-dev@xxxxxxxxxxxxx
> > > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
> > >
> > > On 01/25/2011 05:48 PM, Dietfrid Mali wrote:
> > > > The problem with e.g. G.726 is that Wireshark gives those packets RTP
> > > > type 102 which afaik is an error code ("unknown encoding").
> > >
> > > No, that's your RTP endpoint configured to label these as such. RFC
> > 3550 says:
> > > "A profile MAY specify a default static mapping of payload type codes
> > to payload
> > > formats. Additional payload type codes MAY be defined dynamically
> > through
> > > non-RTP means (see Section 3)."
> > > RFC 1890/RFC 3551 defines the "RTP Profile for Audio and Video
> > Conferences with
> > > Minimal Control", which lists several static payload types. The old
> > RFC lists
> > > G.721 (aka G.726-32), while the new one dropped that one and added
> > references to
> > > G.726 at various bit rate, with a dynamic payload type.
> > > RFC 3550 says in Section 3: "Non-RTP means: Protocols and mechanisms
> > that may be
> > > needed in addition to RTP to provide a usable service. In particular,
> > ..., and
> > > define dynamic mappings between RTP payload type values and the
> > payload formats
> > > they represent for formats that do not have a predefined payload type
> > value."
> > > with reference to Session Description Protocol (SDP)
> > >
> > > So, payload type 102 is a dynamic payload type which has to be given
> > meaning
> > > (through SDP for instance) within the session. In your case Wireshark
> > didn't
> > > pick that up from the trace, hence cannot give you the proper
> > interpretation of
> > > that payload type within that session.
> > >
> > > > I would need to know where and how Wireshark maps dynamic payload types
> > > > (negotiated via SDP) to internal static ones. Above that RFC3551
> > notes that
> > > > static G.726 payload types are obsolete, and afaik there aren't even
> > > > (obsolete) static payload types for all G.726 variants, so Wireshark
> > > > would need to
> > > > take care of that by using some (more or less arbitrary) internal
> > static
> > > > type numbers.
> > >
> > > Yep, that is done by the SDP dissector. It tries to interpret the SDP
> > offer
> > > (should be the answer, but that a whole other story) and create
> > conversations
> > > (see doc/README.developer, section 2.2) for the RTP dissector,
> > feeding it
> > > dynamic payload type information it has learned from the media
> > attributes.
> > >
> > > The RTP dissector does then the heavy lifting on the RTP packets,
> > based on the
> > > information feed in by the SDP dissector.
> > >
> > > > I will do my best to provide a patch once I have fully integrated all
> > > > codecs (currently only G.726-32 has been implemented as proof of
> > > > concept, but since
> > > > this is working adding more is no big deal).
> > >
> > > Just one to get started is fine. Does it integrate into codecs/
> > directory
> > > besides G711a and G711u (and G729 and G723, if you have them)?
> > >
> > > > Getting G.726 to work was a bit of a pain btw because of the weird
> > frame
> > > > sync calculation in rtp_player.c::play_channels() as this function
> > seems to
> > > > assume 1:1 relationships of decoder input and output stream sizes and
> > > > thus simply halves the decoder output batch sizes to determine whether
> > > > frames
> > > > are properly sync'd. This doesn't work for compressed audio. To
> > > > compensate, my decodeG726_32() function doubles the number of bytes
> > returned
> > > > (as it has a 1:2 relationship of input and output buffer sizes). Before
> > > > it did that, lots of silence frames were inserted and half of the audio
> > > > data was
> > > > dropped by the player.
> > >
> > > I'm no sure if I understand you correctly. Working with these decode
> > functions
> > > there is an input buffer with its length as input, and two output
> > parameters,
> > > being the output buffer and it a pointer to store its size. This size
> > of the
> > > output buffer has to be set, by the decoder, to the number of samples
> > in output
> > > buffer. That should be enough, see for instance
> > rtp_player.c:decode_rtp_packet()
> > > the handling of G.279 and G.723.
> > > Be aware that you have to store 16 bit linear samples in the output
> > buffer,
> > > maybe that's your factor 2?
> > >
> > > Thanks,
> > > Jaap
> > >
> > >
> > > >
> > > > Dietfrid
> > > >
> > > > > From: jaap.keuter@xxxxxxxxx
> > > > > Date: Tue, 25 Jan 2011 16:54:25 +0100
> > > > > To: wireshark-dev@xxxxxxxxxxxxx
> > > > > Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark
> > > > >
> > > > > Hi,
> > > > >
> > > > > That would be interesting. Can you put the code in a patch on
> > bugzilla?
> > > > >
> > > > > Can't work on it right now, but would be nice to have.
> > > > >
> > > > > btw: their are already static RTP types assigned for both codecs. The
> > > > dynamic types should come in through protocols like SDP, or a dissector
> > > > preference.
> > > > >
> > > > > Thanks,
> > > > > Jaap
> > > > >
> > > > > Send from my iPhone
> > > > >
> > > > > On 25 jan. 2011, at 16:07, Dietfrid Mali <karx11erx@xxxxxxxxxxx>
> > wrote:
> > > > >
> > > > > > Hi,
> > > > > >
> > > > > > using spandsp, I have added G.722 and G.726 decoders to Wireshark.
> > > > > >
> > > > > > Currently this is a bit of a hack job, particularly regarding
> > > > inclusion of the spandsp lib, and I could need a bit help to properly
> > > > integrate it into Wireshark's automake hell (configure.in).
> > > > > >
> > > > > > There also isn't a proper Wireshark signature for that RTP type (I
> > > > am simply reacting to RTP type 102, which actually is an error
> > code), so
> > > > some help getting this straight and introducing proper codec types
> > would
> > > > be appreciated, too.
> > > > > >
> > > > > >
> > > > > >
>
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