[asterisk-dev] Patch: SIP: choosing common codec and select "free"codec for transcoding

[asterisk]

Very Nice!   This has been a big issues with asterisk.   I look forward
to seeing it in the Bug tracker..

 

Doug

 

 

________________________________

From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Andrey
Sofronov
Sent: Tuesday, January 29, 2008 11:39 AM
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] Patch: SIP: choosing common codec and select
"free"codec for transcoding

 

Hello all!

In my network there are a lot of hardware VoIP gateways and softphones
and I have to make then work. Most of all hardware VoIP gateways support
G.711, G.723 and G.729 codecs and the softphones support G.711, GSM,
iLBC etc codecs. All endpoints are SIP peers. All traffic should pass
through my asterisk server(s). In "vanilla" asterisk there is a
limitation - it is unable to select a common codec for 2 peers using
some algorithm (less bandwidth etc) or if jointcapability = 0 select
codecs for both endpoints that could be transcoded "for free".

1st example:

sip.conf

[210]
disallow=all
allow=alaw,ulaw,g723,g729
...

[220]
disallow=all
allow=alaw,ulaw,g723,g729
...

[255]
disallow=all
allow=gsm
...

210 calls 220:
sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format
Hold     Last Message
10.xx.xx.xx     220         10aa3d3754b  00102/00000  0x8 (alaw)
No       Init: INVITE
10.xx.xx.xx     210         a983a543-52  00101/00939  0x8 (alaw)
No       Rx: INVITE

Works, but alaw is a high-bandwidth codec!

210 calls 255:
sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format
Hold     Last Message
10.xx.xx.xx    255         504c907f5b7  00102/00000  0x2 (gsm)        No
Init: INVITE
10.xx.xx.xx     210         5f84a543-9c  00101/00943  0x8 (alaw)
No       Rx: INVITE

Works here...

2nd example:

sip.conf

[210]
disallow=all
allow=g729,alaw,ulaw,g723
...

[220]
disallow=all
allow=g729,alaw,ulaw,g723
...

[255]
disallow=all
allow=gsm
...

210 calls 220:
sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Format
Hold     Last Message
10.xx.xx.xx     220         572052d8348  00102/00000  0x100 (g729)
No       Tx: ACK
10.xx.xx.xx     210         0486a543-2e  00101/00946  0x100 (g729)
No       Rx: ACK

Good work - g729 pass-through!

210 calls 255:
WARNING[8424]: chan_sip.c:3001 sip_call: No audio format found to offer.
Cancelling call to 255

All codecs are same, but their order was changed.

It's a simple example, just to point a problem.

I'm not a professional C/C++ programmer, but the feature was necessary
so I created a small patch (for asterisk 1.4.17).

In brief:
1) set sip_pvt->prefcodec = 0!
2) Thanks to 1) asterisk continues execution and calls
ast_channel_make_compatible function.
3) In "ast_channel_make_compatible" we recalculate peers' codec
capabilities and sip_pvt->prefs.

Result - the order of the codecs in "allow=" parameter does not matter
anymore. Asterisk will select most efficient codec for pass-through and
if jointcapability = 0 will select free and ecomonic codec for
transcoding.

My patch is not excellent (maybe even awful, but i'm not a programmer
and it works for me fine!) but if anybody get an idea, you can make it
better. Also it would be nice to implement that feature as configuration
parameter.

Please see patch in attachment! Thanks! 

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