[sip-comm-dev] speex and ilbc


#1

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.ProcessEngine@1f18cbe
  Cannot build a flow graph with the customized options:
    Unable to transcode format: LINEAR, 44100.0 Hz, 16-bit, Stereo, LittleEndian, Signed
      outputting to: RAW/RTP

-Ken

For reference, full SIP message for the above:

12:36:26.991 FINE: impl.protocol.sip.ProtocolProviderServiceSipImpl.processResponse().1141 received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.102:57267;branch=z9hG4bK173cb3e60cd2dd24ef364efe94a0cb04;received=74.183.249.212

From: "ken1" <sip:ken1@voipgw.u-strasbg.fr>;tag=858ad3bf

Call-ID: dcecdca5210312068e06e571771d720b@0.0.0.0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:ken@130.79.91.160>
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

···

to: speex/rtp, 8000.0 Hz, 8-bit, Mono, Signed
To: <sip:ken@voipgw.u-strasbg.fr>;tag=as48320023

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#2

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

···

-----Ursprüngliche Nachricht-----
Von: Ken Larson [mailto:kenlars99@users.sourceforge.net]
Gesendet: Donnerstag, 24. April 2008 22:38
An: dev@sip-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.ProcessEngine@1f18cbe
  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.processRespon
se().1141
received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.102:57267;branch=z9hG4bK173cb3e60cd2dd24ef364efe94a0
cb04;received=74.183.249.212
From: "ken1" <sip:ken1@voipgw.u-strasbg.fr>;tag=858ad3bf
To: <sip:ken@voipgw.u-strasbg.fr>;tag=as48320023
Call-ID: dcecdca5210312068e06e571771d720b@0.0.0.0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:ken@130.79.91.160>
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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#3

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 :slight_smile:

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.ProcessEngine@1f18cbe
  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.processResponse().1141
received response=
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.102:57267;branch=z9hG4bK173cb3e60cd2dd24ef364efe94a0cb04;received=74.183.249.212
From: "ken1" <sip:ken1@voipgw.u-strasbg.fr>;tag=858ad3bf
To: <sip:ken@voipgw.u-strasbg.fr>;tag=as48320023
Call-ID: dcecdca5210312068e06e571771d720b@0.0.0.0
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Supported: replaces
Contact: <sip:ken@130.79.91.160>
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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