You could check gsm codec implementation of pjmedia-codec.
I used gsm.c as a template and modified it to intel G729FR successfully.
Only need to care SID frames may be from remote side.

regards,
Gang

On Tue, Jul 8, 2008 at 5:37 AM, Archie Rosenblum archie@bbsti.com wrote:

I don't have files in hand. It was job for my old company.

regards,
Gang

On Tue, Jul 8, 2008 at 10:39 PM, Archie Rosenblum archie@bbsti.com wrote:

On Thu, Aug 14, 2008 at 10:23 PM, Archie Rosenblum archie@bbsti.com wrote:

We're currently integrating Intel IPP codecs for the next release. It's in
the roadmap (http://trac.pjsip.org/repos/roadmap).

Cheers
Benny

Benny Prijono wrote:

This mean that the encoding/decoding routines will done in by the IPP
code itself and not by the soft implementation of the same codec
available inside pjsip.

I am wondering if the very same approach could be done for the APS
available on S60 that's a code that manage the encoding/decoding
directly interfacing with the audio.

Because this would mean providing directly in "hardware" all the codes
(including the tipically non-royalty-free one like AMR) available
natively and for free on the Nokia S60 platform.

It would be a strong enanchment for the pjsip support on the "mobile"
platform environment representing the 76% of the smartphone market!

It could be feasible or not from the pjsip design point of view to have
the APS not only manage the audio but also directly the codec like for
the IPP?

Fabio

Hi Fabio,

AFAIK, S60's APS is more sound device library rather than codec
library, while IPP codecs is purely codecs implementation library. So
I think we cannot use the same approach. As Benny has ever mentioned,
it is possible to use APS codecs in pjmedia, but it will lack some
pjmedia features (i.e: conference and other PCM level media processing
in pjmedia).

It is possible to create S60 native/hardware codec using same approach
as in IPP codecs wrapping as long as there is 'pure' codec API in
S60/Symbian. (There is CMMFCodec class that seems suit this. Not sure
if it is working in S60 3rd ed.)

Regards,
nanang

On 15/08/2008, Fabio Pietrosanti (naif) lists@infosecurity.ch wrote:

Nanang Izzuddin wrote:

That's true but unfortunately on the most widespread devices, like
(Nokia and Samsung) Symbian 9.x S60 3rd edition the CMMFCodec it's only
available trough some SDK plug-in, not on all FPx releases and not on
all phones.

That's the reason why i strongly suggest to use APS, because it's most
widely available on the phones, even where CMMFCodec is not.

The theory:
http://www.symbian.com/developer/techlib/v9.2docs/doc_source/reference/reference-cpp/MMF/CMMFCodecClass.html

The practice :( :
http://wiki.forum.nokia.com/index.php/Codecs
http://wiki.forum.nokia.com/index.php/TSS000457_-_Configuring_CMMFCodec_for_encoding_audio_to_AMR
http://wiki.forum.nokia.com/index.php/Encoding_and_Decoding_using_MMF_based_Codec(CMMFCodec)

Fabio

Dear Fabio,

Do you have an idea on what Nokia E SIP client uses for sound and
codecs?...is it it APS?...

Regards,

Manoj

-----Original Message-----
From: pjsip-bounces@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org]On Behalf Of Fabio Pietrosanti
(naif)
Sent: Thursday, August 21, 2008 9:49 PM
To: pjsip list
Subject: Re: [pjsip] G729

Nanang Izzuddin wrote:

That's true but unfortunately on the most widespread devices, like
(Nokia and Samsung) Symbian 9.x S60 3rd edition the CMMFCodec it's only
available trough some SDK plug-in, not on all FPx releases and not on
all phones.

That's the reason why i strongly suggest to use APS, because it's most
widely available on the phones, even where CMMFCodec is not.

The theory:
http://www.symbian.com/developer/techlib/v9.2docs/doc_source/reference/refer
ence-cpp/MMF/CMMFCodecClass.html

The practice :( :
http://wiki.forum.nokia.com/index.php/Codecs
http://wiki.forum.nokia.com/index.php/TSS000457_-Configuring_CMMFCodec_for
encoding_audio_to_AMR
http://wiki.forum.nokia.com/index.php/Encoding_and_Decoding_using_MMF_based_
Codec(CMMFCodec)

Fabio

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

--
Internal Virus Database is out-of-date.
Checked by AVG.
Version: 7.5.524 / Virus Database: 270.6.3/1612 - Release Date: 8/14/2008
6:03 PM

Manoj Kumar Joshi wrote:

The devil demostrate itself trough the Nokia built-in SIP client.
PJSIP must became even better on S60 platform, and it can!

The built-in SIP clienti use, as far as i remember, AMR-WB at 12kb and
for the sound i don't know because it'a a built-in ROM application and
so maybe could even use CMMFDevSound directly for what we know.

For third party application APS it's the best choice for all S60 Symbian
9.x FP0, FP1, FP2 releases.
From FP3 the new VoIP Audio Services API must be used, but there are
still no FP3 release out-of-the-market and only the first FP2 are
getting out during this months.

Fabio

Hi,

Could you specify more detail about the problem?

Regards,
nanang

On 03/09/2008, Sebastian Groosman Sebastian.Groosman@indicia.nl wrote:

Hi,

I've had the problem that no sound was heard at all with the VoiceAge
G729 libs.
It appeared that for the decoding part I needed a local buffer and
prepend it with a "2".
For the encoding the output buffer and size were the problem.
Both these problems are mostly solved now. (I've included the files)

The only problem I have left, is that the encoded sound sounds metallic.

I followed most of the rules concerning CPU load on windows mobile
(see post on "how to reduce CPU load and improve voice quality on mobile
device FR" and http://trac.pjsip.org/repos/wiki/FAQ#cpu)
Now my G711 and other codes' sound quality is a lot better but the
encoded sound is still metallic.

Any suggestions?

Sebastian

-----Original Message-----
From: pjsip-bounces@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Nanang Izzuddin
Sent: Wednesday, September 03, 2008 4:15 PM
To: pjsip list
Subject: Re: [pjsip] G729

Hi,

Could you specify more detail about the problem?

Regards,
nanang

On 03/09/2008, Sebastian Groosman Sebastian.Groosman@indicia.nl wrote:

smartphone,

VoiceAge

wrote:

VoiceAge or

out.  I

instructions.

It's in

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Hi,
Please try with the following setting first:

• frm_per_pkt = 1
• VAD disabled
  make sure everything ok with this setting, then move on with frm_per_pkt =
  2, if everything ok then deal with VAD enabled.

Some notes about the code:

• in encode(), it seems you ignore the first byte output of VoiceAge G729
  encoder. If that first byte contains frame type info, it shouldn't be
  ignored, especially when VAD is enabled, since the encoder may return SID
  (silence frame) or DTX (discontinuous transmission) or speech frame. If you
  treat SID and DTX frame as speech frame, decoder may fail or produce noise.
• silence detector is unnecessary since G729 already has built-in VAD.
• in encode(), the pcm_in += 2*SAMPLES_PER_FRAME, shouldn't it be just
  pcm_in += SAMPLES_PER_FRAME?
• in decode(), it doesn't seem that SID frame is handled very well: if
  (input->size < L_PACKED_G729A) return PJMEDIA_CODEC_EFRMTOOSHORT;
  Again, if the first byte actually specifies the frame type, instead of
  returning non-PJ_SUCCESS it should be decoded normally (e.g: with first
  byte/frame type = SID)

Regards,
nanang

On Fri, Sep 5, 2008 at 7:30 PM, Sebastian Groosman <
Sebastian.Groosman@indicia.nl> wrote:

Hi Nanang,

Thanks a lot for your input!!

I've handled the items and the sound is crystal clear now J.

There is only one thing I see in the logs... a massive amount of
underflow errors.

Any suggestions on that?

Greets,

Sebastian

11:02:34.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:34.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:34.000  strm002F4254  JB shrinking 1 frame(s), size=12

11:02:34.000  strm002F4254  jb updated(1), prefetch=9, size=14

11:02:34.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:35.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:35.000  strm002F4254  JB shrinking 1 frame(s), size=10

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:35.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:36.000  strm002F4254  JB shrinking 1 frame(s), size=10

11:02:36.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:36.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:36.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:36.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:36.000  strm002F4254  jb updated(2), prefetch=12, size=13

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:38.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:38.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:39.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:39.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:39.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:39.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:39.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:40.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:40.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:41.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:41.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:42.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:42.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:42.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:42.000  strm002F4254  jb updated(1), prefetch=10, size=16

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1
frame

From: pjsip-bounces@lists.pjsip.org
[mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Nanang Izzuddin
Sent: Friday, September 05, 2008 6:06 PM
To: pjsip list
Subject: Re: [pjsip] G729

Hi,

Please try with the following setting first:

• frm_per_pkt = 1

• VAD disabled

make sure everything ok with this setting, then move on with frm_per_pkt
= 2, if everything ok then deal with VAD enabled.

Some notes about the code:

• in encode(), it seems you ignore the first byte output of VoiceAge
  G729 encoder. If that first byte contains frame type info, it shouldn't
  be ignored, especially when VAD is enabled, since the encoder may return
  SID (silence frame) or DTX (discontinuous transmission) or speech frame.
  If you treat SID and DTX frame as speech frame, decoder may fail or
  produce noise.

• silence detector is unnecessary since G729 already has built-in VAD.

• in encode(), the pcm_in += 2*SAMPLES_PER_FRAME, shouldn't it be just
  pcm_in += SAMPLES_PER_FRAME?

• in decode(), it doesn't seem that SID frame is handled very well: if
  (input->size < L_PACKED_G729A) return PJMEDIA_CODEC_EFRMTOOSHORT;

Again, if the first byte actually specifies the frame type, instead of
returning non-PJ_SUCCESS it should be decoded normally (e.g: with first
byte/frame type = SID)

Regards,

nanang

On Fri, Sep 5, 2008 at 7:30 PM, Sebastian Groosman
Sebastian.Groosman@indicia.nl wrote:

Hi,

I've had the problem that no sound was heard at all with the VoiceAge
G729 libs.
It appeared that for the decoding part I needed a local buffer and
prepend it with a "2".
For the encoding the output buffer and size were the problem.
Both these problems are mostly solved now. (I've included the files)

The only problem I have left, is that the encoded sound sounds metallic.

I followed most of the rules concerning CPU load on windows mobile
(see post on "how to reduce CPU load and improve voice quality on mobile
device FR" and http://trac.pjsip.org/repos/wiki/FAQ#cpu)
Now my G711 and other codes' sound quality is a lot better but the
encoded sound is still metallic.

Any suggestions?

Sebastian

-----Original Message-----
From: pjsip-bounces@lists.pjsip.org

[mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Nanang Izzuddin
Sent: Wednesday, September 03, 2008 4:15 PM
To: pjsip list
Subject: Re: [pjsip] G729

Hi,

Could you specify more detail about the problem?

Regards,
nanang

On 03/09/2008, Sebastian Groosman Sebastian.Groosman@indicia.nl wrote:

smartphone,

VoiceAge

wrote:

VoiceAge or

out.  I

instructions.

It's in

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Sebastian,

When you've completed you project, would you share your files?  Perhaps,
your G729 implementation could become part of the base code?

Take care,

Archie

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Sebastian Groosman
Sent: Monday, September 08, 2008 5:14 AM
To: pjsip list
Subject: Re: [pjsip] G729

Hi Nanang,

Thanks a lot for your input!!

I've handled the items and the sound is crystal clear now :-).

There is only one thing I see in the logs. a massive amount of underflow
errors.

Any suggestions on that?

Greets,

Sebastian

11:02:34.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:34.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:34.000  strm002F4254  JB shrinking 1 frame(s), size=12

11:02:34.000  strm002F4254  jb updated(1), prefetch=9, size=14

11:02:34.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:35.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:35.000  strm002F4254  JB shrinking 1 frame(s), size=10

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:35.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:35.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:36.000  strm002F4254  JB shrinking 1 frame(s), size=10

11:02:36.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:36.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:36.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:36.000  strm002F4254  JB shrinking 1 frame(s), size=11

11:02:36.000  strm002F4254  jb updated(2), prefetch=12, size=13

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:37.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:38.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:38.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:38.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:39.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:39.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:39.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:39.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:39.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:40.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:40.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:40.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:41.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:41.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:41.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:42.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:42.000  strm002F4254  JB shrinking 1 frame(s), size=13

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:42.000  strm002F4254  JB shrinking 1 frame(s), size=14

11:02:42.000  strm002F4254  jb updated(1), prefetch=10, size=16

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

11:02:42.000  Master/sound  Underflow, buf_cnt=0, will generate 1 frame

From: pjsip-bounces@lists.pjsip.org [mailto:pjsip-bounces@lists.pjsip.org]
On Behalf Of Nanang Izzuddin
Sent: Friday, September 05, 2008 6:06 PM
To: pjsip list
Subject: Re: [pjsip] G729

Hi,

Please try with the following setting first:

• frm_per_pkt = 1

• VAD disabled

make sure everything ok with this setting, then move on with frm_per_pkt =
2, if everything ok then deal with VAD enabled.

Some notes about the code:

• in encode(), it seems you ignore the first byte output of VoiceAge G729
  encoder. If that first byte contains frame type info, it shouldn't be
  ignored, especially when VAD is enabled, since the encoder may return SID
  (silence frame) or DTX (discontinuous transmission) or speech frame. If you
  treat SID and DTX frame as speech frame, decoder may fail or produce noise.

• silence detector is unnecessary since G729 already has built-in VAD.

• in encode(), the pcm_in += 2*SAMPLES_PER_FRAME, shouldn't it be just
  pcm_in += SAMPLES_PER_FRAME?

• in decode(), it doesn't seem that SID frame is handled very well: if
  (input->size < L_PACKED_G729A) return PJMEDIA_CODEC_EFRMTOOSHORT;

Again, if the first byte actually specifies the frame type, instead of
returning non-PJ_SUCCESS it should be decoded normally (e.g: with first
byte/frame type = SID)

Regards,

nanang

On Fri, Sep 5, 2008 at 7:30 PM, Sebastian Groosman
Sebastian.Groosman@indicia.nl wrote:

Hi,

I've had the problem that no sound was heard at all with the VoiceAge
G729 libs.
It appeared that for the decoding part I needed a local buffer and
prepend it with a "2".
For the encoding the output buffer and size were the problem.
Both these problems are mostly solved now. (I've included the files)

The only problem I have left, is that the encoded sound sounds metallic.

I followed most of the rules concerning CPU load on windows mobile
(see post on "how to reduce CPU load and improve voice quality on mobile
device FR" and http://trac.pjsip.org/repos/wiki/FAQ#cpu)
Now my G711 and other codes' sound quality is a lot better but the
encoded sound is still metallic.

Any suggestions?

Sebastian

-----Original Message-----
From: pjsip-bounces@lists.pjsip.org

[mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Nanang Izzuddin
Sent: Wednesday, September 03, 2008 4:15 PM
To: pjsip list
Subject: Re: [pjsip] G729

Hi,

Could you specify more detail about the problem?

Regards,
nanang

On 03/09/2008, Sebastian Groosman Sebastian.Groosman@indicia.nl wrote:

smartphone,

VoiceAge

wrote:

VoiceAge or

out.  I

instructions.

It's in

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

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