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.
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.
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:
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
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
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]