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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