PJSIP without audio device
I'm trying to set up an embedded device (currently without audio hardware) to use PJSIP. The final device WILL have audio hardware but, for now, we just want to test whether PJSIP will build and function correctly in our environment.
So the setup we desire is a client application which will make a call to a server, read some audio data from a file, then send that via PJSIP to the server. The server will receive the call and write the audio data to another file. Then it will be a (relatively) simple matter of comparing the two files to see if they are the same content (by listening to them since I'm not sure the audio transmission will be a full bit-for-bit copy).
Keep in mind the function names or structures may look slightly different to what you're used to - we're actually using a SWIG-generated thin C# layer over pjsua2.
We have the server written to the point where it's awaiting an incoming call so that's not a problem (yet). It kicks up and waits on port 5060 for an incoming call.
The client follows the same logic as the server in terms of initialising (setting up ConnectionSettings and Account and creating delegates to capture PJSIP events). However, rather than waiting for an incoming call as the server does, it instead calls:
call = pjsip.Account.Call("127.0.0.1", 5060)
to try and contact the server. At this point, we see an error output to the command line (the timestamped lines are out of PJSIP itself, other lines are our log messages):
=====
== Starting the VOIP demo client
== Instantiating connection settings
== Instantiating Pjsip object
02:01:16.185 os_core_unix.c !pjlib 2.4.5 for POSIX initialized
02:01:16.188 sip_endpoint.c .Creating endpoint instance...
02:01:16.188 pjlib .select() I/O Queue created (0x13b1318)
02:01:16.188 sip_endpoint.c .Module "mod-msg-print" registered
02:01:16.188 sip_transport. .Transport manager created.
02:01:16.189 pjsua_core.c .PJSUA state changed: NULL --> CREATED
02:01:16.190 sip_endpoint.c .Module "mod-pjsua-log" registered
02:01:16.190 sip_endpoint.c .Module "mod-tsx-layer" registered
02:01:16.190 sip_endpoint.c .Module "mod-stateful-util" registered
02:01:16.190 sip_endpoint.c .Module "mod-ua" registered
02:01:16.190 sip_endpoint.c .Module "mod-100rel" registered
02:01:16.190 sip_endpoint.c .Module "mod-pjsua" registered
02:01:16.191 sip_endpoint.c .Module "mod-invite" registered
02:01:16.204 alsa_dev.c ..ALSA driver found 0 devices
02:01:16.204 alsa_dev.c ..ALSA initialized
02:01:16.204 pjlib ..select() I/O Queue created (0x13b2134)
02:01:16.225 sip_endpoint.c .Module "mod-evsub" registered
02:01:16.225 sip_endpoint.c .Module "mod-presence" registered
02:01:16.225 sip_endpoint.c .Module "mod-mwi" registered
02:01:16.225 sip_endpoint.c .Module "mod-refer" registered
02:01:16.225 sip_endpoint.c .Module "mod-pjsua-pres" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-im" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-options" registered
02:01:16.226 pjsua_core.c .1 SIP worker threads created
02:01:16.226 pjsua_core.c .pjsua version 2.4.5 for Linux-3.14.28/armv7l/glibc-2.21 initialized
02:01:16.226 pjsua_core.c .PJSUA state changed: CREATED --> INIT
02:01:16.227 pjsua_core.c PJSUA state changed: INIT --> STARTING
02:01:16.227 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
02:01:16.227 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
PJSUA2 Started
== Initialising Pjsip object
02:01:16.235 pjsua_core.c SIP UDP socket reachable at 169.254.113.10:5070
02:01:16.235 udp0x13e7e18 SIP UDP transport started, published address is 169.254.113.10:5070
02:01:16.261 pjsua_acc.c Adding account: id=sip:127.0.0.1:5070
02:01:16.261 pjsua_acc.c .Account sip:127.0.0.1:5070 added with id 0
== Adding delegates
== Status update, callActive: False, callConnected: False
== PJSUA2 started
== Making call
***** DtiCall constructor
***** DtiCall constructor done
== Status update, callActive: True, callConnected: False
02:01:16.289 pjsua_call.c Making call with acc #0 to sip:127.0.0.1:5060
02:01:16.289 pjsua_aud.c .Set sound device: capture=-1, playback=-2
02:01:16.289 pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
02:01:16.289 call.cpp pjsua_call_make_call(acc.getId(), &pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) (status=420006) [../src/pjsua2/call.cpp:485]
Unhandled Exception:
System.ApplicationException: C++ pj::Error exception thrown
at PjsipSharp.Call.makeCall (System.String dst_uri, PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0
at DtiVoipSimple.DtiAccount.Call (System.String ipAddress, UInt32 port) [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Run () [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Main (System.String[] args) [0x00000] in <filename unknown>:0
[ERROR] FATAL UNHANDLED EXCEPTION: System.ApplicationException: C++ pj::Error exception thrown
at PjsipSharp.Call.makeCall (System.String dst_uri, PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0
at DtiVoipSimple.DtiAccount.Call (System.String ipAddress, UInt32 port) [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Run () [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Main (System.String[] args) [0x00000] in <filename unknown>:0
As you can see at the bottom, the call is failing because there are apparently no audio devices available. We know that there are none available, the hardware for that is not yet ready and dmesg lists the ALSA device list as empty. I suspect the server may also complain similarly when it tries to accept the call but we haven't got that far yet.
What I'm asking is if it's possible to use PJSIP without audio hardware? While we are using this only for initial testing, it appears to me this is actually a desirable feature should anyone have a system in which the computer itself is supposed to send audio over PJSIP, without an actual speaker orating the message.
Allan Chandler | Software Engineer
DTI Group Ltd | Transit Security & Surveillance
31 Affleck Road, Perth Airport, WA 6105, AU
P | F | allan.chandler@dti.com.au
Visit our website www.dti.com.auhttp://www.dti.com.au
The information contained in this email is confidential. If you receive this email in error, please inform DTI Group Ltd via the above contact details. If you are not the intended recipient, you may not use or disclose the information contained in this email or attachments.
Hi Allen, PJSIP provides a null audio device which provides timing for audio media. This is what you want. - Bill
Sent from my iPhone
On Jul 18, 2016, at 4:32 AM, Allan Chandler allan.chandler@dti.com.au wrote:
I'm trying to set up an embedded device (currently without audio hardware) to use PJSIP. The final device WILL have audio hardware but, for now, we just want to test whether PJSIP will build and function correctly in our environment.
So the setup we desire is a client application which will make a call to a server, read some audio data from a file, then send that via PJSIP to the server. The server will receive the call and write the audio data to another file. Then it will be a (relatively) simple matter of comparing the two files to see if they are the same content (by listening to them since I'm not sure the audio transmission will be a full bit-for-bit copy).
Keep in mind the function names or structures may look slightly different to what you're used to - we're actually using a SWIG-generated thin C# layer over pjsua2.
We have the server written to the point where it's awaiting an incoming call so that's not a problem (yet). It kicks up and waits on port 5060 for an incoming call.
The client follows the same logic as the server in terms of initialising (setting up ConnectionSettings and Account and creating delegates to capture PJSIP events). However, rather than waiting for an incoming call as the server does, it instead calls:
call = pjsip.Account.Call("127.0.0.1", 5060)
to try and contact the server. At this point, we see an error output to the command line (the timestamped lines are out of PJSIP itself, other lines are our log messages):
=====
== Starting the VOIP demo client
== Instantiating connection settings
== Instantiating Pjsip object
02:01:16.185 os_core_unix.c !pjlib 2.4.5 for POSIX initialized
02:01:16.188 sip_endpoint.c .Creating endpoint instance...
02:01:16.188 pjlib .select() I/O Queue created (0x13b1318)
02:01:16.188 sip_endpoint.c .Module "mod-msg-print" registered
02:01:16.188 sip_transport. .Transport manager created.
02:01:16.189 pjsua_core.c .PJSUA state changed: NULL --> CREATED
02:01:16.190 sip_endpoint.c .Module "mod-pjsua-log" registered
02:01:16.190 sip_endpoint.c .Module "mod-tsx-layer" registered
02:01:16.190 sip_endpoint.c .Module "mod-stateful-util" registered
02:01:16.190 sip_endpoint.c .Module "mod-ua" registered
02:01:16.190 sip_endpoint.c .Module "mod-100rel" registered
02:01:16.190 sip_endpoint.c .Module "mod-pjsua" registered
02:01:16.191 sip_endpoint.c .Module "mod-invite" registered
02:01:16.204 alsa_dev.c ..ALSA driver found 0 devices
02:01:16.204 alsa_dev.c ..ALSA initialized
02:01:16.204 pjlib ..select() I/O Queue created (0x13b2134)
02:01:16.225 sip_endpoint.c .Module "mod-evsub" registered
02:01:16.225 sip_endpoint.c .Module "mod-presence" registered
02:01:16.225 sip_endpoint.c .Module "mod-mwi" registered
02:01:16.225 sip_endpoint.c .Module "mod-refer" registered
02:01:16.225 sip_endpoint.c .Module "mod-pjsua-pres" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-im" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-options" registered
02:01:16.226 pjsua_core.c .1 SIP worker threads created
02:01:16.226 pjsua_core.c .pjsua version 2.4.5 for Linux-3.14.28/armv7l/glibc-2.21 initialized
02:01:16.226 pjsua_core.c .PJSUA state changed: CREATED --> INIT
02:01:16.227 pjsua_core.c PJSUA state changed: INIT --> STARTING
02:01:16.227 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
02:01:16.227 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
PJSUA2 Started
== Initialising Pjsip object
02:01:16.235 pjsua_core.c SIP UDP socket reachable at 169.254.113.10:5070
02:01:16.235 udp0x13e7e18 SIP UDP transport started, published address is 169.254.113.10:5070
02:01:16.261 pjsua_acc.c Adding account: id=sip:127.0.0.1:5070
02:01:16.261 pjsua_acc.c .Account sip:127.0.0.1:5070 added with id 0
== Adding delegates
== Status update, callActive: False, callConnected: False
== PJSUA2 started
== Making call
***** DtiCall constructor
***** DtiCall constructor done
== Status update, callActive: True, callConnected: False
02:01:16.289 pjsua_call.c Making call with acc #0 to sip:127.0.0.1:5060
02:01:16.289 pjsua_aud.c .Set sound device: capture=-1, playback=-2
02:01:16.289 pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
02:01:16.289 call.cpp pjsua_call_make_call(acc.getId(), &pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) (status=420006) [../src/pjsua2/call.cpp:485]
Unhandled Exception:
System.ApplicationException: C++ pj::Error exception thrown
at PjsipSharp.Call.makeCall (System.String dst_uri, PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0
at DtiVoipSimple.DtiAccount.Call (System.String ipAddress, UInt32 port) [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Run () [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Main (System.String[] args) [0x00000] in <filename unknown>:0
[ERROR] FATAL UNHANDLED EXCEPTION: System.ApplicationException: C++ pj::Error exception thrown
at PjsipSharp.Call.makeCall (System.String dst_uri, PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0
at DtiVoipSimple.DtiAccount.Call (System.String ipAddress, UInt32 port) [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Run () [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Main (System.String[] args) [0x00000] in <filename unknown>:0
As you can see at the bottom, the call is failing because there are apparently no audio devices available. We know that there are none available, the hardware for that is not yet ready and dmesg lists the ALSA device list as empty. I suspect the server may also complain similarly when it tries to accept the call but we haven't got that far yet.
What I'm asking is if it's possible to use PJSIP without audio hardware? While we are using this only for initial testing, it appears to me this is actually a desirable feature should anyone have a system in which the computer itself is supposed to send audio over PJSIP, without an actual speaker orating the message.
Allan Chandler | Software Engineer
DTI Group Ltd | Transit Security & Surveillance
31 Affleck Road, Perth Airport, WA 6105, AU
P | F | allan.chandler@dti.com.au
Visit our website www.dti.com.au
The information contained in this email is confidential. If you receive this email in error, please inform DTI Group Ltd via the above contact details. If you are not the intended recipient, you may not use or disclose the information contained in this email or attachments.
Visit our blog: http://blog.pjsip.org
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Thanks for that, Bill.
I've now successfully gotten a client application to talk to a server (in terms of establishing a call), the only main problem being that the server crashes on exit. I'm not too concerned about that yet since it's almost certainly to do with deferred disposal, something the PJSUA2 docs warn about when using non-native bindings. I'll fix it at some point but it's not high on the priority list at the moment.
What I now need to do is to be able to read an audio file (since the hardware has no ALSA devices) on the client and send it over PJSIP to the server, which will then write that data to another file for comparison.
I've found the AudiMediaPlayer stuff but that appears to just be for sending the file to a local sound device whereas I need to send it over the wire to the server.
I suspect from an earlier comment that the NULL sound device just provides timing, that I will need to somehow hook into the timer and send audio data to the call based on that.
So, two questions.
1/ How do I get the client to send audio data to the server? Reading the file is easy, I just need to know how to feed that into the call.
2/ How does this audio show up at the server (e.g., which callback/delegate), so I can write it to a file?
Cheers,
Al.
From: pjsip [mailto:pjsip-bounces@lists.pjsip.org] On Behalf Of Bill Gardner
Sent: Monday, 18 July 2016 2:31 PM
To: pjsip list
Subject: Re: [pjsip] PJSIP without audio device
Hi Allen, PJSIP provides a null audio device which provides timing for audio media. This is what you want. - Bill
Sent from my iPhone
On Jul 18, 2016, at 4:32 AM, Allan Chandler <allan.chandler@dti.com.aumailto:allan.chandler@dti.com.au> wrote:
I'm trying to set up an embedded device (currently without audio hardware) to use PJSIP. The final device WILL have audio hardware but, for now, we just want to test whether PJSIP will build and function correctly in our environment.
So the setup we desire is a client application which will make a call to a server, read some audio data from a file, then send that via PJSIP to the server. The server will receive the call and write the audio data to another file. Then it will be a (relatively) simple matter of comparing the two files to see if they are the same content (by listening to them since I'm not sure the audio transmission will be a full bit-for-bit copy).
Keep in mind the function names or structures may look slightly different to what you're used to - we're actually using a SWIG-generated thin C# layer over pjsua2.
We have the server written to the point where it's awaiting an incoming call so that's not a problem (yet). It kicks up and waits on port 5060 for an incoming call.
The client follows the same logic as the server in terms of initialising (setting up ConnectionSettings and Account and creating delegates to capture PJSIP events). However, rather than waiting for an incoming call as the server does, it instead calls:
call = pjsip.Account.Call("127.0.0.1", 5060)
to try and contact the server. At this point, we see an error output to the command line (the timestamped lines are out of PJSIP itself, other lines are our log messages):
=====
== Starting the VOIP demo client
== Instantiating connection settings
== Instantiating Pjsip object
02:01:16.185 os_core_unix.c !pjlib 2.4.5 for POSIX initialized
02:01:16.188 sip_endpoint.c .Creating endpoint instance...
02:01:16.188 pjlib .select() I/O Queue created (0x13b1318)
02:01:16.188 sip_endpoint.c .Module "mod-msg-print" registered
02:01:16.188 sip_transport. .Transport manager created.
02:01:16.189 pjsua_core.c .PJSUA state changed: NULL --> CREATED
02:01:16.190 sip_endpoint.c .Module "mod-pjsua-log" registered
02:01:16.190 sip_endpoint.c .Module "mod-tsx-layer" registered
02:01:16.190 sip_endpoint.c .Module "mod-stateful-util" registered
02:01:16.190 sip_endpoint.c .Module "mod-ua" registered
02:01:16.190 sip_endpoint.c .Module "mod-100rel" registered
02:01:16.190 sip_endpoint.c .Module "mod-pjsua" registered
02:01:16.191 sip_endpoint.c .Module "mod-invite" registered
02:01:16.204 alsa_dev.c ..ALSA driver found 0 devices
02:01:16.204 alsa_dev.c ..ALSA initialized
02:01:16.204 pjlib ..select() I/O Queue created (0x13b2134)
02:01:16.225 sip_endpoint.c .Module "mod-evsub" registered
02:01:16.225 sip_endpoint.c .Module "mod-presence" registered
02:01:16.225 sip_endpoint.c .Module "mod-mwi" registered
02:01:16.225 sip_endpoint.c .Module "mod-refer" registered
02:01:16.225 sip_endpoint.c .Module "mod-pjsua-pres" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-im" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-options" registered
02:01:16.226 pjsua_core.c .1 SIP worker threads created
02:01:16.226 pjsua_core.c .pjsua version 2.4.5 for Linux-3.14.28/armv7l/glibc-2.21 initialized
02:01:16.226 pjsua_core.c .PJSUA state changed: CREATED --> INIT
02:01:16.227 pjsua_core.c PJSUA state changed: INIT --> STARTING
02:01:16.227 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
02:01:16.227 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
PJSUA2 Started
== Initialising Pjsip object
02:01:16.235 pjsua_core.c SIP UDP socket reachable at 169.254.113.10:5070
02:01:16.235 udp0x13e7e18 SIP UDP transport started, published address is 169.254.113.10:5070
02:01:16.261 pjsua_acc.c Adding account: id=sip:127.0.0.1:5070
02:01:16.261 pjsua_acc.c .Account sip:127.0.0.1:5070 added with id 0
== Adding delegates
== Status update, callActive: False, callConnected: False
== PJSUA2 started
== Making call
***** DtiCall constructor
***** DtiCall constructor done
== Status update, callActive: True, callConnected: False
02:01:16.289 pjsua_call.c Making call with acc #0 to sip:127.0.0.1:5060
02:01:16.289 pjsua_aud.c .Set sound device: capture=-1, playback=-2
02:01:16.289 pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006]
02:01:16.289 call.cpp pjsua_call_make_call(acc.getId(), &pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) (status=420006) [../src/pjsua2/call.cpp:485]
Unhandled Exception:
System.ApplicationException: C++ pj::Error exception thrown
at PjsipSharp.Call.makeCall (System.String dst_uri, PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0
at DtiVoipSimple.DtiAccount.Call (System.String ipAddress, UInt32 port) [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Run () [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Main (System.String[] args) [0x00000] in <filename unknown>:0
[ERROR] FATAL UNHANDLED EXCEPTION: System.ApplicationException: C++ pj::Error exception thrown
at PjsipSharp.Call.makeCall (System.String dst_uri, PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0
at DtiVoipSimple.DtiAccount.Call (System.String ipAddress, UInt32 port) [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Run () [0x00000] in <filename unknown>:0
at DtiVoipSimple.MainClass.Main (System.String[] args) [0x00000] in <filename unknown>:0
As you can see at the bottom, the call is failing because there are apparently no audio devices available. We know that there are none available, the hardware for that is not yet ready and dmesg lists the ALSA device list as empty. I suspect the server may also complain similarly when it tries to accept the call but we haven't got that far yet.
What I'm asking is if it's possible to use PJSIP without audio hardware? While we are using this only for initial testing, it appears to me this is actually a desirable feature should anyone have a system in which the computer itself is supposed to send audio over PJSIP, without an actual speaker orating the message.
Allan Chandler | Software Engineer
DTI Group Ltd | Transit Security & Surveillance
31 Affleck Road, Perth Airport, WA 6105, AU
P | F | allan.chandler@dti.com.aumailto:allan.chandler@dti.com.au
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Hi Al,
I've never used pjsua2 api, but this is trivially done using pjsua.
You'd create a wav_player, add it to the conference, and then connect
its port to the call port. Then media will stream from the wav file. You
can see the code for pjsua_player_create for example. The other user
agent (the server in your case) will receive the audio as if it came
from a mic recording. Note however that the transmission is not
reliable, there may be packets missing or reconstructed using PLC, and
the audio is encoded/decoded using whatever codec is selected for the
call. If the server is also implemented using pjsip it is easy to
capture the received audio into a wav file using a wav_writer.
Bill
On 7/19/2016 9:40 PM, Allan Chandler wrote:
Thanks for that, Bill.
I've now successfully gotten a client application to talk to a server
(in terms of establishing a call), the only main problem being that
the server crashes on exit. I'm not too concerned about that yet since
it's almost certainly to do with deferred disposal, something the
PJSUA2 docs warn about when using non-native bindings. I'll fix it at
some point but it's not high on the priority list at the moment.
What I now need to do is to be able to read an audio file (since the
hardware has no ALSA devices) on the client and send it over PJSIP to
the server, which will then write that data to another file for
comparison.
I've found the AudiMediaPlayer stuff but that appears to just be for
sending the file to a local sound device whereas I need to send it
over the wire to the server.
I suspect from an earlier comment that the NULL sound device just
provides timing, that I will need to somehow hook into the timer and
send audio data to the call based on that.
So, two questions.
1/ How do I get the client to send audio data to the server? Reading
the file is easy, I just need to know how to feed that into the call.
2/ How does this audio show up at the server (e.g., which
callback/delegate), so I can write it to a file?
Cheers,
Al.
*From:*pjsip [mailto:pjsip-bounces@lists.pjsip.org] *On Behalf Of
*Bill Gardner
Sent: Monday, 18 July 2016 2:31 PM
To: pjsip list
Subject: Re: [pjsip] PJSIP without audio device
Hi Allen, PJSIP provides a null audio device which provides timing for
audio media. This is what you want. - Bill
Sent from my iPhone
On Jul 18, 2016, at 4:32 AM, Allan Chandler <allan.chandler@dti.com.au
mailto:allan.chandler@dti.com.au> wrote:
I'm trying to set up an embedded device (currently without audio
hardware) to use PJSIP. The final device WILL have audio hardware
but, for now, we just want to test whether PJSIP will build and
function correctly in our environment.
So the setup we desire is a client application which will make a
call to a server, read some audio data from a file, then send that
via PJSIP to the server. The server will receive the call and
write the audio data to another file. Then it will be a
(relatively) simple matter of comparing the two files to see if
they are the same content (by listening to them since I'm not sure
the audio transmission will be a full bit-for-bit copy).
Keep in mind the function names or structures may look slightly
different to what you're used to - we're actually using a
SWIG-generated thin C# layer over pjsua2.
We have the server written to the point where it's awaiting an
incoming call so that's not a problem (yet). It kicks up and waits
on port 5060 for an incoming call.
The client follows the same logic as the server in terms of
initialising (setting up ConnectionSettings and Account and
creating delegates to capture PJSIP events). However, rather than
waiting for an incoming call as the server does, it instead calls:
call = pjsip.Account.Call("127.0.0.1", 5060)
to try and contact the server. At this point, we see an error
output to the command line (the timestamped lines are out of PJSIP
itself, other lines are our log messages):
=====
== Starting the VOIP demo client
== Instantiating connection settings
== Instantiating Pjsip object
02:01:16.185 os_core_unix.c !pjlib 2.4.5 for POSIX initialized
02:01:16.188 sip_endpoint.c .Creating endpoint instance...
02:01:16.188 pjlib .select() I/O Queue created (0x13b1318)
02:01:16.188 sip_endpoint.c .Module "mod-msg-print" registered
02:01:16.188 sip_transport. .Transport manager created.
02:01:16.189 pjsua_core.c .PJSUA state changed: NULL --> CREATED
02:01:16.190 sip_endpoint.c .Module "mod-pjsua-log" registered
02:01:16.190 sip_endpoint.c .Module "mod-tsx-layer" registered
02:01:16.190 sip_endpoint.c .Module "mod-stateful-util" registered
02:01:16.190 sip_endpoint.c .Module "mod-ua" registered
02:01:16.190 sip_endpoint.c .Module "mod-100rel" registered
02:01:16.190 sip_endpoint.c .Module "mod-pjsua" registered
02:01:16.191 sip_endpoint.c .Module "mod-invite" registered
02:01:16.204 alsa_dev.c ..ALSA driver found 0 devices
02:01:16.204 alsa_dev.c ..ALSA initialized
02:01:16.204 pjlib ..select() I/O Queue created (0x13b2134)
02:01:16.225 sip_endpoint.c .Module "mod-evsub" registered
02:01:16.225 sip_endpoint.c .Module "mod-presence" registered
02:01:16.225 sip_endpoint.c .Module "mod-mwi" registered
02:01:16.225 sip_endpoint.c .Module "mod-refer" registered
02:01:16.225 sip_endpoint.c .Module "mod-pjsua-pres" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-im" registered
02:01:16.226 sip_endpoint.c .Module "mod-pjsua-options" registered
02:01:16.226 pjsua_core.c .1 SIP worker threads created
02:01:16.226 pjsua_core.c .pjsua version 2.4.5 for
Linux-3.14.28/armv7l/glibc-2.21 initialized
02:01:16.226 pjsua_core.c .PJSUA state changed: CREATED --> INIT
02:01:16.227 pjsua_core.c PJSUA state changed: INIT --> STARTING
02:01:16.227 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
02:01:16.227 pjsua_core.c .PJSUA state changed: STARTING -->
RUNNING
PJSUA2 Started
== Initialising Pjsip object
02:01:16.235 pjsua_core.c SIP UDP socket reachable at
169.254.113.10:5070
02:01:16.235 udp0x13e7e18 SIP UDP transport started, published
address is 169.254.113.10:5070
02:01:16.261 pjsua_acc.c Adding account: id=sip:127.0.0.1:5070
02:01:16.261 pjsua_acc.c .Account sip:127.0.0.1:5070 added
with id 0
== Adding delegates
== Status update, callActive: False, callConnected: False
== PJSUA2 started
== Making call
***** DtiCall constructor
***** DtiCall constructor done
== Status update, callActive: True, callConnected: False
02:01:16.289 pjsua_call.c Making call with acc #0 to
sip:127.0.0.1:5060
02:01:16.289 pjsua_aud.c .Set sound device: capture=-1,
playback=-2
*02:01:16.289 pjsua_aud.c ..Error retrieving default audio
device parameters: Unable to find default audio device
(PJMEDIA_EAUD_NODEFDEV) [status=420006]*
*02:01:16.289 call.cpp pjsua_call_make_call(acc.getId(),
&pj_dst_uri, param.p_opt, this, param.p_msg_data, &id) error:
Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV)
(status=420006) [../src/pjsua2/call.cpp:485]*
**
*Unhandled Exception:*
*System.ApplicationException: C++ pj::Error exception thrown*
* at PjsipSharp.Call.makeCall (System.String dst_uri,
PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0*
* at DtiVoipSimple.DtiAccount.Call (System.String ipAddress,
UInt32 port) [0x00000] in <filename unknown>:0*
* at DtiVoipSimple.MainClass.Run () [0x00000] in <filename
unknown>:0*
* at DtiVoipSimple.MainClass.Main (System.String[] args)
[0x00000] in <filename unknown>:0*
*[ERROR] FATAL UNHANDLED EXCEPTION: System.ApplicationException:
C++ pj::Error exception thrown*
* at PjsipSharp.Call.makeCall (System.String dst_uri,
PjsipSharp.CallOpParam prm) [0x00000] in <filename unknown>:0*
* at DtiVoipSimple.DtiAccount.Call (System.String ipAddress,
UInt32 port) [0x00000] in <filename unknown>:0*
* at DtiVoipSimple.MainClass.Run () [0x00000] in <filename
unknown>:0*
* at DtiVoipSimple.MainClass.Main (System.String[] args)
[0x00000] in <filename unknown>:0*
=====
As you can see at the bottom, the call is failing because there
are apparently no audio devices available. We know that there are
none available, the hardware for that is not yet ready and dmesg
lists the ALSA device list as empty. I suspect the server may also
complain similarly when it tries to accept the call but we haven't
got that far yet.
What I'm asking is if it's possible to use PJSIP without audio
hardware? While we are using this only for initial testing, it
appears to me this is actually a desirable feature should anyone
have a system in which the computer itself is supposed to send
audio over PJSIP, without an actual speaker orating the message.
*Allan Chandler*| Software Engineer
*DTI Group Ltd*| *Transit Security & Surveillance*
31 Affleck Road, Perth Airport, WA 6105, AU
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