pjsip over app protocol?

RL
Richard Langly
Wed, Aug 5, 2009 2:49 AM

I'm not well versed w/ voip, and am looking for a voip lib that will
allow me to easily grab the voice stream and wrap it into my own
application layer protocol, get the voice stream from point a to piont
b, and unwrap it at the other end.

So, basically make a point-to-point voip call, yet ... use a voice
stream to help me see how well my app layer protocol performs.

Is pjsip something I could easily do this with? If so, any pointers?
If not, any recommendations?

Any help much appreciated.

I'm not well versed w/ voip, and am looking for a voip lib that will allow me to easily grab the voice stream and wrap it into my own application layer protocol, get the voice stream from point a to piont b, and unwrap it at the other end. So, basically make a point-to-point voip call, yet ... use a voice stream to help me see how well my app layer protocol performs. Is pjsip something I could easily do this with? If so, any pointers? If not, any recommendations? Any help much appreciated.
SO
Shayne O'Neill
Wed, Aug 5, 2009 3:18 AM

I'm not well versed w/ voip, and am looking for a voip lib that will
allow me to easily grab the voice stream and wrap it into my own
application layer protocol, get the voice stream from point a to piont
b, and unwrap it at the other end.

So, basically make a point-to-point voip call, yet ... use a voice
stream to help me see how well my app layer protocol performs.

Is pjsip something I could easily do this with? If so, any pointers?
If not, any recommendations?

Any help much appreciated.

Ok, I might be misunderstanding you, but tell me if this helps;-

Have a look at some of the test apps like the sine wave generator.
PJSip has kind of a patch bay architecture that lets you define end
points (like audio devices, sip calls, tone generators) and lets you
arbitrarily patch them into each other.

The sine generator is instructive in this case. It should be fairly
straight forward to adapt that code into a generic end point template
you can use to provide your own data stream.

You basically create an end point, register it, then use a call to
patch it in via the conference bridge. Its a rather powerful arangement.

On 05/08/2009, at 10:49 AM, Richard Langly wrote:

---==
Shayne O'Neill Development
Mobile, Web and Business process integration.
shayne.oneill@gmail.com 0400247091
Ask me about how Alfresco can help your business grow.

> I'm not well versed w/ voip, and am looking for a voip lib that will > allow me to easily grab the voice stream and wrap it into my own > application layer protocol, get the voice stream from point a to piont > b, and unwrap it at the other end. > > So, basically make a point-to-point voip call, yet ... use a voice > stream to help me see how well my app layer protocol performs. > > Is pjsip something I could easily do this with? If so, any pointers? > If not, any recommendations? > > Any help much appreciated. Ok, I might be misunderstanding you, but tell me if this helps;- Have a look at some of the test apps like the sine wave generator. PJSip has kind of a patch bay architecture that lets you define end points (like audio devices, sip calls, tone generators) and lets you arbitrarily patch them into each other. The sine generator is instructive in this case. It should be fairly straight forward to adapt that code into a generic end point template you can use to provide your own data stream. You basically create an end point, register it, then use a call to patch it in via the conference bridge. Its a rather powerful arangement. On 05/08/2009, at 10:49 AM, Richard Langly wrote: > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org =================================== Shayne O'Neill Development Mobile, Web and Business process integration. shayne.oneill@gmail.com 0400247091 Ask me about how Alfresco can help your business grow.
RL
Richard Langly
Wed, Aug 5, 2009 3:47 AM

Ugh, dumb question time, I'm grep'ing for anything 'sine wave
generator' and don't know which example your talking about here.

On Tue, Aug 4, 2009 at 10:18 PM, Shayne O'Neillshayne.oneill@gmail.com wrote:

I'm not well versed w/ voip, and am looking for a voip lib that will
allow me to easily grab the voice stream and wrap it into my own
application layer protocol, get the voice stream from point a to piont
b, and unwrap it at the other end.

So, basically make a point-to-point voip call, yet ... use a voice
stream to help me see how well my app layer protocol performs.

Is pjsip something I could easily do this with? If so, any pointers?
If not, any recommendations?

Any help much appreciated.

Ok, I might be misunderstanding you, but tell me if this helps;-

Have a look at some of the test apps like the sine wave generator. PJSip has
kind of a patch bay architecture that lets you define end points (like audio
devices, sip calls, tone generators) and lets you arbitrarily patch them
into each other.

The sine generator is instructive in this case. It should be fairly straight
forward to adapt that code into a generic end point template you can use to
provide your own data stream.

You basically create an end point, register it, then use a call to patch it
in via the conference bridge. Its a rather powerful arangement.

On 05/08/2009, at 10:49 AM, Richard Langly wrote:

---==
Shayne O'Neill Development
Mobile, Web and Business process integration.
shayne.oneill@gmail.com 0400247091
Ask me about how Alfresco can help your business grow.


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

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

Ugh, dumb question time, I'm grep'ing for anything 'sine wave generator' and don't know which example your talking about here. On Tue, Aug 4, 2009 at 10:18 PM, Shayne O'Neill<shayne.oneill@gmail.com> wrote: >> I'm not well versed w/ voip, and am looking for a voip lib that will >> allow me to easily grab the voice stream and wrap it into my own >> application layer protocol, get the voice stream from point a to piont >> b, and unwrap it at the other end. >> >> So, basically make a point-to-point voip call, yet ... use a voice >> stream to help me see how well my app layer protocol performs. >> >> Is pjsip something I could easily do this with? If so, any pointers? >> If not, any recommendations? >> >> Any help much appreciated. > > Ok, I might be misunderstanding you, but tell me if this helps;- > > Have a look at some of the test apps like the sine wave generator. PJSip has > kind of a patch bay architecture that lets you define end points (like audio > devices, sip calls, tone generators) and lets you arbitrarily patch them > into each other. > > The sine generator is instructive in this case. It should be fairly straight > forward to adapt that code into a generic end point template you can use to > provide your own data stream. > > You basically create an end point, register it, then use a call to patch it > in via the conference bridge. Its a rather powerful arangement. > > On 05/08/2009, at 10:49 AM, Richard Langly wrote: >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip@lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > =================================== > Shayne O'Neill Development > Mobile, Web and Business process integration. > shayne.oneill@gmail.com 0400247091 > Ask me about how Alfresco can help your business grow. > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
SO
Shayne O'Neill
Wed, Aug 5, 2009 4:02 AM

Have a look  in pjsip-apps / src / samples

Sent from my iPhone

On 05/08/2009, at 11:47 AM, Richard Langly <richard.ringo.langly@gmail.com

wrote:

Ugh, dumb question time, I'm grep'ing for anything 'sine wave
generator' and don't know which example your talking about here.

On Tue, Aug 4, 2009 at 10:18 PM, Shayne O'Neill<shayne.oneill@gmail.com

wrote:

I'm not well versed w/ voip, and am looking for a voip lib that will
allow me to easily grab the voice stream and wrap it into my own
application layer protocol, get the voice stream from point a to
piont
b, and unwrap it at the other end.

So, basically make a point-to-point voip call, yet ... use a voice
stream to help me see how well my app layer protocol performs.

Is pjsip something I could easily do this with? If so, any pointers?
If not, any recommendations?

Any help much appreciated.

Ok, I might be misunderstanding you, but tell me if this helps;-

Have a look at some of the test apps like the sine wave generator.
PJSip has
kind of a patch bay architecture that lets you define end points
(like audio
devices, sip calls, tone generators) and lets you arbitrarily patch
them
into each other.

The sine generator is instructive in this case. It should be fairly
straight
forward to adapt that code into a generic end point template you
can use to
provide your own data stream.

You basically create an end point, register it, then use a call to
patch it
in via the conference bridge. Its a rather powerful arangement.

On 05/08/2009, at 10:49 AM, Richard Langly wrote:

---==
Shayne O'Neill Development
Mobile, Web and Business process integration.
shayne.oneill@gmail.com 0400247091
Ask me about how Alfresco can help your business grow.


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

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

Have a look in pjsip-apps / src / samples Sent from my iPhone On 05/08/2009, at 11:47 AM, Richard Langly <richard.ringo.langly@gmail.com > wrote: > Ugh, dumb question time, I'm grep'ing for anything 'sine wave > generator' and don't know which example your talking about here. > > On Tue, Aug 4, 2009 at 10:18 PM, Shayne O'Neill<shayne.oneill@gmail.com > > wrote: >>> I'm not well versed w/ voip, and am looking for a voip lib that will >>> allow me to easily grab the voice stream and wrap it into my own >>> application layer protocol, get the voice stream from point a to >>> piont >>> b, and unwrap it at the other end. >>> >>> So, basically make a point-to-point voip call, yet ... use a voice >>> stream to help me see how well my app layer protocol performs. >>> >>> Is pjsip something I could easily do this with? If so, any pointers? >>> If not, any recommendations? >>> >>> Any help much appreciated. >> >> Ok, I might be misunderstanding you, but tell me if this helps;- >> >> Have a look at some of the test apps like the sine wave generator. >> PJSip has >> kind of a patch bay architecture that lets you define end points >> (like audio >> devices, sip calls, tone generators) and lets you arbitrarily patch >> them >> into each other. >> >> The sine generator is instructive in this case. It should be fairly >> straight >> forward to adapt that code into a generic end point template you >> can use to >> provide your own data stream. >> >> You basically create an end point, register it, then use a call to >> patch it >> in via the conference bridge. Its a rather powerful arangement. >> >> On 05/08/2009, at 10:49 AM, Richard Langly wrote: >>> >>> _______________________________________________ >>> Visit our blog: http://blog.pjsip.org >>> >>> pjsip mailing list >>> pjsip@lists.pjsip.org >>> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> =================================== >> Shayne O'Neill Development >> Mobile, Web and Business process integration. >> shayne.oneill@gmail.com 0400247091 >> Ask me about how Alfresco can help your business grow. >> >> >> _______________________________________________ >> 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
RL
Richard Langly
Wed, Aug 5, 2009 4:39 AM

I was able to get samples that play wav files and other to work, but
the playsine app didn't work for me. I get the following error. I'm
running gentoo on a quad-core. Any thoughts as to what I'm doing wrong
here? I don't see a config file anywhere's that I'm missing.

/pjproject-1.3/pjsip-apps/bin/samples/x86_64-unknown-linux-gnu$ playsine 2
23:37:23.128 os_core_unix.c  pjlib 1.3 for POSIX initialized
23:37:23.198      pa_dev.c  PortAudio sound library initialized, status=0
23:37:23.198      pa_dev.c  PortAudio host api count=2
23:37:23.198      pa_dev.c  Sound device count=14
23:37:23.198          pjlib  select() I/O Queue created (0x2288ed8)
23:37:23.199      pa_dev.c  PA message: Expression
'SetApproximateSampleRate( pcm, hwParams, sr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1291

23:37:23.199      pa_dev.c  PA message: Expression
'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams,
self->primeBuffers, hwParamsPlayback, &realSr )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1865

23:37:23.199      pa_dev.c  PA message: Expression
'PaAlsaStream_Configure( stream, inputParameters, outputParameters,
sampleRate, framesPerBuffer, &inputLatency, &outputLatency,
&hostBufferSizeMode )' failed in
'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1986

Unable to open sound device: Invalid sample rate [code=469996]

I was able to get samples that play wav files and other to work, but the playsine app didn't work for me. I get the following error. I'm running gentoo on a quad-core. Any thoughts as to what I'm doing wrong here? I don't see a config file anywhere's that I'm missing. /pjproject-1.3/pjsip-apps/bin/samples/x86_64-unknown-linux-gnu$ playsine 2 23:37:23.128 os_core_unix.c pjlib 1.3 for POSIX initialized 23:37:23.198 pa_dev.c PortAudio sound library initialized, status=0 23:37:23.198 pa_dev.c PortAudio host api count=2 23:37:23.198 pa_dev.c Sound device count=14 23:37:23.198 pjlib select() I/O Queue created (0x2288ed8) 23:37:23.199 pa_dev.c PA message: Expression 'SetApproximateSampleRate( pcm, hwParams, sr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1291 23:37:23.199 pa_dev.c PA message: Expression 'PaAlsaStreamComponent_InitialConfigure( &self->playback, outParams, self->primeBuffers, hwParamsPlayback, &realSr )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1865 23:37:23.199 pa_dev.c PA message: Expression 'PaAlsaStream_Configure( stream, inputParameters, outputParameters, sampleRate, framesPerBuffer, &inputLatency, &outputLatency, &hostBufferSizeMode )' failed in 'src/../../../portaudio/src/hostapi/alsa/pa_linux_alsa.c', line: 1986 Unable to open sound device: Invalid sample rate [code=469996]