speex and ilbc

To what extent do speex and ilbc actually work with SC?  If
they haven't 
been tested, that is fine, I just need to know, I think I
can get them 
to work with some code changes, but I just don't want to
break anything.

I'm now calling from one SC to another, I want to test speex
and ilbc.  
I'm using JMF, not FMJ.

Observing the SIP traffic, I see:

a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000

but nothing for the other formats.  Should they be there? 

In any case, if I force the format to be speex (regardless
of the sdp), 
I get an exception:

Failed to build a graph for the given custom options.
Failed to realize: com.sun.media.ProcessEngine1f18cbe
  Cannot build a flow graph with the customized options:
    Unable to transcode format: LINEAR, 44100.0 Hz, 16-bit,
Stereo, 
LittleEndian, Signed
      to: speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed
      outputting to: RAW/RTP


-Ken

For reference, full SIP message for the above:

12:36:26.991 FINE: 
impl.protocol.sip.ProtocolProviderServiceSipImpl.processResp
onse().1141 
received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.0.102:57267;branch=z9hG4bK173cb3e60cd2dd24ef364efe94
a0cb04;received=74.183.249.212
From: "ken1" <sip:ken1voipgw.u-strasbg.fr>;tag=858ad3bf
To: <sip:kenvoipgw.u-strasbg.fr>;tag=as48320023
Call-ID: dcecdca5210312068e06e571771d720b0.0.0.0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:ken130.79.91.160>
Content-Type: application/sdp
Content-Length: 347

v=0
o=root 3326 3326 IN IP4 130.79.91.160
s=session
c=IN IP4 130.79.91.160
b=CT:384
t=0 0
m=audio 18646 RTP/AVP 0 8 3 4
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18654 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

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Hi Ken!

We are using the ILBC codec for SIP Communicator to SIP
Communicator calls
since 6 months now in production systems and I've heard no
complaints. It
pretty much worked out of the box, so I haven't had a look
at the internals.
We're not using the latest trunk, but a version from about 9
months ago, so
some things (like the SIP content negotiation) might have
changed.

By the way, I'd like to say a big Thank You to the
developers who have made
this possible.

Cheers
Michael Koch

> -----Urspr�ngliche Nachricht-----
> Von: Ken Larson [mailto:kenlars99users.sourceforge.net] 
> Gesendet: Donnerstag, 24. April 2008 22:38
> An: devsip-communicator.dev.java.net
> Betreff: [sip-comm-dev] speex and ilbc
> 
> To what extent do speex and ilbc actually work with SC?
 If 
> they haven't 
> been tested, that is fine, I just need to know, I think
I can 
> get them 
> to work with some code changes, but I just don't want
to 
> break anything.
> 
> I'm now calling from one SC to another, I want to test
speex 
> and ilbc.  
> I'm using JMF, not FMJ.
> 
> Observing the SIP traffic, I see:
> 
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:4 G723/8000
> 
> but nothing for the other formats.  Should they be
there? 
> 
> In any case, if I force the format to be speex
(regardless of 
> the sdp), 
> I get an exception:
> 
> Failed to build a graph for the given custom options.
> Failed to realize: com.sun.media.ProcessEngine1f18cbe
>   Cannot build a flow graph with the customized
options:
>     Unable to transcode format: LINEAR, 44100.0 Hz,
16-bit, Stereo, 
> LittleEndian, Signed
>       to: speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed
>       outputting to: RAW/RTP
> 
> 
> -Ken
> 
> For reference, full SIP message for the above:
> 
> 12:36:26.991 FINE: 
>
impl.protocol.sip.ProtocolProviderServiceSipImpl.processResp
on
> se().1141 
> received response=
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 
>
192.168.0.102:57267;branch=z9hG4bK173cb3e60cd2dd24ef364efe94
a0
> cb04;received=74.183.249.212
> From: "ken1" <sip:ken1voipgw.u-strasbg.fr>;tag=858ad3bf
> To: <sip:kenvoipgw.u-strasbg.fr>;tag=as48320023
> Call-ID: dcecdca5210312068e06e571771d720b0.0.0.0
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
> Supported: replaces
> Contact: <sip:ken130.79.91.160>
> Content-Type: application/sdp
> Content-Length: 347
> 
> v=0
> o=root 3326 3326 IN IP4 130.79.91.160
> s=session
> c=IN IP4 130.79.91.160
> b=CT:384
> t=0 0
> m=audio 18646 RTP/AVP 0 8 3 4
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:4 G723/8000
> a=fmtp:4 annexa=no
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 18654 RTP/AVP 34
> a=rtpmap:34 H263/90000
> a=sendrecv
> 
>
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---------
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> dev-helpsip-communicator.dev.java.net
> 

Hi Ken,

Sorry for not replying earlier. I had a few things to handle
here so I
accumulated quite a baclog. Anyways, things are calmer now


So, as you have noticed ilbc and speex are only half
implemented. Up
till now we have been testing them by forcing the codec
through the
asterisk conversation but I won't be surprised if that has
stopped
working at some point since we don't use them regularly.

In other words - if you see things that are plain wrong in
there then
that's simply because they are wrong and there's no other
reason. So no
worries, I doubt you'd break anything in there.

Cheers
Emil

Ken Larson написа:
> To what extent do speex and ilbc actually work with SC?
 If they haven't 
> been tested, that is fine, I just need to know, I think
I can get them 
> to work with some code changes, but I just don't want
to break anything.
> 
> I'm now calling from one SC to another, I want to test
speex and ilbc.  
> I'm using JMF, not FMJ.
> 
> Observing the SIP traffic, I see:
> 
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:4 G723/8000
> 
> but nothing for the other formats.  Should they be
there? 
> 
> In any case, if I force the format to be speex
(regardless of the sdp), 
> I get an exception:
> 
> Failed to build a graph for the given custom options.
> Failed to realize: com.sun.media.ProcessEngine1f18cbe
>   Cannot build a flow graph with the customized
options:
>     Unable to transcode format: LINEAR, 44100.0 Hz,
16-bit, Stereo, 
> LittleEndian, Signed
>       to: speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed
>       outputting to: RAW/RTP
> 
> 
> -Ken
> 
> For reference, full SIP message for the above:
> 
> 12:36:26.991 FINE: 
>
impl.protocol.sip.ProtocolProviderServiceSipImpl.processResp
onse().1141 
> received response=
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 
>
192.168.0.102:57267;branch=z9hG4bK173cb3e60cd2dd24ef364efe94
a0cb04;received=74.183.249.212
> From: "ken1" <sip:ken1voipgw.u-strasbg.fr>;tag=858ad3bf
> To: <sip:kenvoipgw.u-strasbg.fr>;tag=as48320023
> Call-ID: dcecdca5210312068e06e571771d720b0.0.0.0
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow:
INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
> Supported: replaces
> Contact: <sip:ken130.79.91.160>
> Content-Type: application/sdp
> Content-Length: 347
> 
> v=0
> o=root 3326 3326 IN IP4 130.79.91.160
> s=session
> c=IN IP4 130.79.91.160
> b=CT:384
> t=0 0
> m=audio 18646 RTP/AVP 0 8 3 4
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:4 G723/8000
> a=fmtp:4 annexa=no
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 18654 RTP/AVP 34
> a=rtpmap:34 H263/90000
> a=sendrecv
> 
>
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> 
> 



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