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Re: G.722 and G.726 decoders for Wireshark

From: "Kukosa, Tomas" <tomas.kukosa () siemens-enterprise com>
Date: Thu, 27 Jan 2011 15:05:45 +0100
Hi,

for the first step I would stay with 8kHz audio format and rtp_palyer can be extended for other rates in the next step.

Current implementation works just for assigned payload types (not dynamic) and codec has to be registered with name 
from rtp_payload_type_short_vals table (see epan/dissectors/packet-rtp.c)

As G.726 uses dynamic payload type it could be registered with encoding name (G726-32) and RTP player needs to get 
information about dynamic payload type from RTP conversation (it is not implemented now).

________________________________

From: wireshark-dev-bounces () wireshark org [mailto:wireshark-dev-bounces () wireshark org] On Behalf Of Dietfrid Mali
Sent: Thursday, January 27, 2011 11:22 AM
To: wireshark-dev () wireshark org
Subject: Re: [Wireshark-dev] G.722 and G.726 decoders for Wireshark


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 () siemens-enterprise com
To: wireshark-dev () wireshark org
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 () wireshark org [mailto:wireshark-dev-bounces () wireshark org] 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 () xs4all nl
To: wireshark-dev () wireshark org
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 () xs4all nl
Date: Tue, 25 Jan 2011 16:54:25 +0100
To: wireshark-dev () wireshark org
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 () hotmail com>

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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