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PJSIP FAQ

Here you can find answers to some of the most frequently asked questions about PJSIP. If you have a question not answered on this page, you can ask it on the PJSIP mailing list.

General Questions
Build and Installation Issues
1. I'm having problem building PJSIP, please help!
2. I'm getting "Unable to open DSound.h error", please help!
Integration Issues
Audio Questions
Video Questions
SIP Questions
Performance
1. How can I reduce the CPU usage/maximize the performance of my application?
2. How can I configure pjsip to serve thousands of calls?
Footprint
1. How can I reduce the footprint and heap usage of my application?
Windows Specific Questions
1. How can I use PJSIP in .NET applications?
Windows Mobile Specific Questions
1. How can I build PJSIP for WinCE/Windows Mobile?
Linux/uC-Linux Specific Questions
1. How can I build PJSIP for Linux/uC-Linux?
MacOS X Specific Questions
1. How can I build PJSIP for MacOS X?
Symbian Specific Questions
1. How can I build PJSIP for Symbian?
Python
1. How can I use PJSIP in Python?

General Questions

Why is PJSIP licensed as GPL, and not (LGPL|MPL|BSD|choose your OSS license here)?

Basically we agree with FSF on this issue. Quoting the GPL FAQ:

Why should I use the GNU GPL rather than other free software licenses?

Using the GNU GPL will require that all the released improved versions be free software. This means you can avoid the risk of having to compete with a proprietary modified version of your own work.

We don't want people to take PJSIP, mess it up (erm, improve it), and keep the improvements as proprietary code. On the contrary, we want everybody to enjoy PJSIP and all its improvements, and the only way to make sure of this is by releasing PJSIP as GPL.

What about the "viral" nature of the GPL?

People often think that using GPL-ed software means that other software linked with the GPL software have to be GPL too, hence GPL is considered as viral. That is not exactly true, especially with PJSIP. When linking with PJSIP, only your application (that is based on PJSIP) has to be GPL-ed, since it is considered as derived work. Other libraries linked with PJSIP don't have to be GPL, because we specifically allow linking PJSIP with third party libraries, as long as they are listed in Third Party Software that are allowed to be linked with PJSIP.

Can I develop closed source products with PJSIP?

It depends. We use the standard GPL v2 or later for PJSIP, and GPL does allow using GPL-ed code for closed source development, as long as the resulting product is not redistributed (for example, it is only used for internal purpose). Please see GPL FAQ for more information about what you can/can't do with GPL software.

Alternatively, we can discuss alternative licensing for PJSIP, please contact support@… for details.

What SIP products are compatible with PJSIP?

Basically as PJSIP is based on IETF standards (SIP, RTP/RTCP, STUN, ICE, etc.), so it should be compatible with other standard based products. We and the community are constantly using PJSIP with other open source products such as:

We have also heard people successfully done interoperability tests with commercial SIP servers or products, but we don't specificly maintain list of these products. Basically the principle stays the same, as long as they follow the standards, we are confident that PJSIP should be able to communicate with them.

Where can I find the latest release of PJSIP?

The latest PJSIP is always the SVN trunk version. You can find information on how to retrieve PJSIP from SVN in the PJSIP Download and Getting Started page.

Build and Installation Issues

I'm having problem building PJSIP, please help!

You should check the Getting Started page.

I'm getting "Unable to open DSound.h error", please help!

You should really check the Getting Started page first.

Integration Issues

Can I use the SIP stack with my own media stack?

Sure. The SIP stack (pjsip) and media stack (pjmedia) is independent from one another, and they are only integrated at the higher level user agent library (pjsua-lib). Please see the library architecture diagram on PJSIP Documentation page for reference.

So yes, you can use the SIP stack (pjsip) without having to use pjmedia, as long as you don't use pjsua-lib.

Can I use the media stack with my own SIP stack?

Yes. Please see above.

What platforms are PJSIP known to run on?

PJSIP runs on many platforms, embedded or non-embedded. Please see the platforms list that it currently supports in Platform features page. Apart from the platforms that we maintain ourselves, the community have also been constantly finding new platforms that PJSIP can run on. For now we don't maintain list of platforms that the community have ported PJSIP to, but you can search for them in the mailing list archive.

How can I port PJSIP to platform XYZ?

PJSIP has been designed for ultra-portability, and we have ported PJSIP to exotic platform such as Symbian (long time ago, PJSIP has also been ported to Linux kernel). People even have ported PJSIP to even more exotic platforms such as Nintendo DS and Texas Instrument (TI) DSP. So there is a good chance that PJSIP will be port-able to your platform.

You can start from the Porting page. For additional information, there is also information on porting PJSIP to Symbian, to emphasize that even on difficult platform like this, PJSIP is still port-able.

Audio Questions

I'm having problems with no/poor audio (quality), please help!

We have created a Wiki page dedicated for troubleshooting all sorts of audio problems, please follow the instructions in http://www.pjsip.org/trac/wiki/sound-problems. If this does not solve your problem, please consult the PJSIP mailing list

I'm having problems with echo, please help

Please see above.

Why does PJSIP refuse to send DTMF to remote?

PJSIP will only send RFC 2833 DTMF to remote if remote has indicated its capability to accept RFC 2833 events in its SDP. This is done by putting this line in the SDP:

a=rtpmap:101 telephone-event/8000

Without receiving this capability indication, PJSIP will refuse to send RFC 2833 event, and a call to pjsua_call_dial_dtmf() or pjmedia_session_dial_dtmf() or pjmedia_stream_dial_dtmf() will return error code PJMEDIA_RTP_EREMNORFC2833. In this case, you can send the DTMF tone inband instead (see below).

How can I send inband DTMF tones?

It's quite easy with PJSUA-LIB API:

Once you get the call established, create an instance of Tone Generator.
Register this tone generator to pjsua's conference bridge, by calling pjsua_conf_add_port().
Connect the tone generator to the call, with pjsua_conf_connect().
Now instruct the tone generator to play some DTMF digits (pjmedia_tonegen_play_digits()). The digits then will be streamed to the call, and remote endpoint should receive the DTMF tone inband.

Outgoing RTP transmissions are not timed equally/properly. Why?

In PJMEDIA default setup, media flow is normally triggered by the clock from the audio device. We made this kind of design to ensure that the audio device always gets fed whenever it needs to be fed, and also since this design would work best for DSP devices, where audio flow would be triggered by some kind of (soft) IRQs thus it should provide the best timing source for audio flow (and without needing to have multi-threading capability).

Unfortunately, not all audio devices provide good timing. Especially in PC world, and also with some uC-Linux based development boards that only support OSS, it is very common to have sound cards that can't provide reliable timing. On these platforms, you will notice that audio frames come in burst rather than one by one and spaced equally. So with 20ms ptime for example, rather than having one frame every 20ms, these devices would give PJMEDIA three or four frames every 60ms or 80ms. Since RTP packets are transmitted as soon as audio frame is available from the sound card, this would cause PJMEDIA to transmit RTP packets at (what looks like) irregular interval.

In my opinion, this should be fine, as the remote endpoint should be able to accommodate this with its jitter buffer.

If you don't like this, and you'd rather want PJMEDIA to transmit RTP packets at good interval, you can install a master clock port between sound device and conference bridge, so that the master port will drive the media clock instead. A master clock port uses an internal thread to drive the media flow, so it should provide better timing on most platforms.

The steps to install master port between sound device and conference bridge are as follows:

Create a splitter/combiner (splitcomb) port.
Create a reverse phase port on the splitcomb (pjmedia_splitcomb_create_rev_channel()).
Create the sound device port as usual.
Connect the sound device port to the splitcomb (use pjmedia_snd_port_connect()).
Create a master clock port, and specify the splitcomb's reverse channel as the upstream port, and the conference bridge as the downstream port.
Start the master port.

For example, if you normally connect the sound device to the conference bridge (the default setup in pjsua-lib) with the code below:

void connect_conf_bridge_to_snd_dev(pj_pool_t *pool, pjmedia_port *conf)
{
    pjmedia_snd_port *snd;

    pjmedia_snd_port_create(..., &snd);
    pjmedia_snd_port_connect(snd, conf);
}

Then the change required to install master clock between sound device and conference bridge would be something like this:

void connect_conf_bridge_to_snd_dev2(pj_pool_t *pool, pjmedia_port *conf)
{
    pjmedia_port *splitcomb, *rev;
    pjmedia_snd_port *snd;
    pjmedia_master_port *m;

    pjmedia_splitcomb_create(pool, CLOCK_RATE, 1, SAMPLES_PER_FRAME,
                             BITS, 0, &splitcomb);
    pjmedia_splitcomb_create_rev_channel(pool, splitcomb, 0, 0, &rev);
    
    pjmedia_snd_port_create(..., &snd);
    pjmedia_snd_port_connect(snd, splitcomb);

    pjmedia_master_port_create(pool, rev, conf, 0, &m);
    pjmedia_master_port_start(m);
}

Does PJSIP support G.723 or G.729 codec?

Yes and no.

No, because there is no ready to use G.723/G.729 codec implementation in PJMEDIA. We specifically don't include G.723/G.729 support in our code because G.723/G.729 are both patented and royalty based codecs, so we are quite nervous with the possibility that some lawyers may contact us should we include them in PJMEDIA. So our decision is to include only free and open source codec in PJMEDIA.

But yes, because you can always add your own codec in PJMEDIA. Please see Adding a new codec question below for more info.

How to add new codec to PJMEDIA?

First of all, read the Codec Framework documentation. Then, the easiest is to take other codec source file in pjmedia-codec directory (such as gsm.c), and replace GSM specific function calls with the functions that are provided by your codec library.

Once you're finished, you should end up with just two public APIs exported by your codec: the initialization function, and the deinitialization function.

Then register your codec to PJMEDIA in your application by calling the codec initialization function. After this, your codec should be picked up automagically by the rest of PJMEDIA framework.

How can I manipulate audio samples directly?

In PJMEDIA, audio frames are sent back and forth between what is called media port (pjmedia_port). So to be able to peek or manipulate audio frames, you need to implement your own media port.

Implementing media port should be easy. Basically you just need to implement these:

Create your media port structure.
Fill in the media port information to describe your media port (like, the name, clock rate, bits per sample, etc.). Use pjmedia_port_info_init() to initialize the port into.
Implement get_frame() callback (of the pjmedia_port) if your media port is a source (that is, your media port feed audio frames to other media ports).
Implement put_frame() callback (of the pjmedia_port) if your media port is a sink (that is, other media ports may feed audio frames to your media port).
Implement on_destroy(), if you need to reclaim resources when the media port is destroyed.

There are many media port sample implementations in PJMEDIA.

For source only media ports, samples include:

playsine.c from the pjsip-apps/samples directory.
mem_player.c from pjmedia (media port to playback audio from a buffer).
wav_player.c from pjmedia (media port to playback audio from WAVE file).
tonegen.c from pjmedia (media port to generate sine waves/DTMF/multi-frequency tones).

For sink only media ports, samples include:

mem_capture.c from pjmedia (media port to save audio to a buffer).
wav_writer.c from pjmedia (media port to save audio to a WAVE file).

For media ports that manipulates audio and provide both sink and source callbacks:

resample_port.c from pjmedia (to convert sampling rate)
echo_port.c from pjmedia (the AEC)

I always get "Bad RTP pt" error. Why?

From our experience, this can be caused by one of these:

Remote endpoint has agreed to use one codec in the SDP negotiation, but it sends RTP with different codec. This had happened with some old version of a popular but not open source softphone (don't want to name it to protect the innocent). When this happens, the "Bad RTP pt" error message will be printed continuously in the log or screen (basically for every RTP packet!). The remedy in this case was to upgrade that softphone to a new version.
Remote endpoint is sending a comfort noise packet. When this happens, the error message is not printed as often as the other case (maybe once in every few seconds).

When you encounter this problem, and if upgrading the softphone doesn't solve the problem, you can report this to PJSIP mailing list. When reporting, please include the error log and the complete INVITE message and the 200/OK response (containing the SDP), so that we can analyze which endpoint is behaving badly.

Does PJSIP support SRTP or ZRTP? If no, how can I implement them?

Currently PJMEDIA doesn't support SRTP nor ZRTP, although this has been on our TODO list for sometime now. Fortunately, implementing them is not too difficult, since media transport in PJMEDIA is separated from the stream.

Have a look at:

transport_udp.c, the normal UDP media transport, and
transport_ice.c, ICE media transport.

There, it's quite easy to see where RTP/RTCP packets are received (that is in on_rx_rtp()/on_rx_rtcp() respectively) or sent (in transport_send_rtp()/transport_send_rtcp()). So these are the locations where you should call the decryption and encryption function of SRTP/ZRTP.

Alternatively, there is a cleaner way to plug in SRTP/ZRTP functionality without changing existing code. As mentioned earlier, media transport is separated from stream. They are only attached to each other in application code (or in pjsua-lib). So then one can build an adapter, which is plugged between the media transport and the stream to do the SRTP/ZRTP stuffs. To the stream, this adapter will look like a media transport, and to existing media transport, this adapter will look like a stream. The benefit of this approach is we can use the same adapter for both kind of media transports, that is the UDP and ICE media transport.

Video Questions

Does PJSIP support video?

Unfortunately no at the moment. Supporting video has been in our TODO list for long time (more than a year!), but so far we haven't had enough time and commitment to do it.

When can I expect PJSIP to support video?

We're not sure.

How can I add video support in PJSIP?

Since video stream is independent from audio stream, the simplest "integration" would be to not integrate video with pjmedia at all. In other words, the application would have to manage everything related to video (capture, render, codec, and RTP) by itself, separate from audio stream which is managed by pjmedia. Maybe you can use the media transports supplied by pjmedia, and the RTP module to pack/unpack RTP packets, to help you up a little bit, but apart from these, basically you'll have to do everything on your own.

Also some processing needs to be done on the SDP to support video. So basically, for outgoing SDP, application has to add video information in the SDP after the SDP is created, and it has to parse video related information in the incoming SDP by itself.

And perhaps if synchronization between audio and video is needed, the application then will need to peek the RTP packet coming to the audio stream. This can be done perhaps by modifying the media transport, or by installing media transport adapter as described in SRTP/ZRTP question above.

SIP Questions

Does PJSIP support RFC XYZ?

The best way is to look at the PJSIP Features page and see if that particular feature is supported by PJSIP.

I'm having problem when calling XYZ, please help me!

Please report your problem to PJSIP mailing list. In order for us to help you, please provide the following information:

Please describe in detail what are you trying to accomplish and how do you configure your equipments (for example, do you use proxy? What kind of proxy?).
It would be best to try to reproduce the problem with pjsua, so that we can also try to reproduce your problem.
In any case, please provide the complete log file. If you use pjsua, you can configure it to write log file with --log-file option. If you use your own application that is based on pjsua-lib, you can write the log by setting pjsua_logging_config.log_filename field.

I cannot login/REGISTER to my server. It complains about authentication error.

Most likely this is caused by wrong credential in your configuration. The remedy depends on what error was reported by PJSIP.

For PJSIP_ENOCREDENTIAL error:

This error is caused by the realm specified in the credential doesn't match the realm challenged by the server in the 401/407 response. If you use PJSIP version 0.7-trunk or PJSIP version 0.7.1 or later, you can put wildcard ("*") as the realm to make PJSIP respond to any realms challenged by the server. If you use older PJSIP, you have to match the realm in the credential with the realm in the challenge. The realm normally would be equal to the domain name, but it doesn't have to. Asterisk, for example, always set the realm to "asterisk".

For PJSIP_EFAILEDCREDENTIAL error:

If you encounter this error, most likely it was caused by a wrong credential. Check if the username and password combination is correct.

With ICE enabled, INVITE packet can exceed MTU/1500 bytes. How can I reduce the message size?

Please see ticket #342 for info. Basically there are few settings to configure to make PJSIP sends smaller packet:

Configure PJSIP to send compact form of SIP headers, by setting PJSIP_ENCODE_SHORT_HNAME macro to 1 in your config_site.h. This will reduce SIP message size by approximately 50 bytes.
Suppress the inclusion of Allow header in outgoing requests, by setting PJSIP_INCLUDE_ALLOW_HDR_IN_DLG macro to 0 in config_site.h. This will reduce SIP message size by approximately 86 bytes.
Suppress the inclusion of SDP rtpmap attribute for static payload types, by setting PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT macro to 0 in config_site.h. Note that this setting should not cause bad effects for communication, since SDP rtpmap attributes for static payload types are optional. This will reduce SIP message size by approximately 65 bytes.
Disable RTCP (advertisement) in SDP, by setting PJMEDIA_ADVERTISE_RTCP macro to 0 in config_site.h. When RTCP is disabled, no RTCP packets will be sent or received, and this will cause some RTCP TX statistics (including RTT report) to be unavailable. Other RTCP statistics such as RX statistics, as well as number of TX packets, will still be available since these values are generated locally. Disabling RTCP will reduce SIP message size by approximately 235 bytes for ICE with three candidates.
Last resort, disable some unused network interfaces in your system to reduce the number of ICE candidates advertised in SDP. On Windows for example, it's quite common to have some Loopback network interface, and disabling it doesn't seem to cause (too much) harm.

How can I instantiate multiple PJSIP stacks in my application?

You don't need to! This may be necessary with other SIP stacks, but PJSIP inherently supports multiple identities (or accounts) throughout the libraries, so one instance of the stack can be used to perform multiple, separate registration and invite sessions.

Performance

How can I reduce the CPU usage/maximize the performance of my application?

There are few configuration settings to tweak to reduce the CPU usage of the application or to produce the best performance out of pjsip:

Echo canceller. The software AEC probably is the most CPU intensive module in PJSIP. To reduce the CPU usage, you can shorten the EC tail length to lower value (the pjsua_media_config.ec_tail_len setting), or you can even disable it altogether by setting pjsua_media_config.ec_tail_len to zero.
Codec. Use low complexity codec such as pcmu or pcma. When you use pcmu or pcma, make sure pjmedia chooses the table based implementation, by setting PJMEDIA_HAS_ALAW_ULAW_TABLE macro to 1 (the default is normally enabled, but it may worth double-checking).
Avoid resampling. Resampling is a CPU intensive process, thus you should avoid it if you can, by choosing uniform clock rate for all media components (sound device, conference bridge, codecs, WAV files, etc.).
Logging. The default logging level is 5, which provides verbose information just in case some debugging is needed. When absolute performance is needed, you can decrease the logging verbosity level to 3 so that only vital information is displayed. This can be done by, either at compile time by setting PJ_LOG_MAX_LEVEL macro (in config_site.h as usual), or at run-time by calling pj_log_set_level().
Threads. Use the optimum number of worker threads in your application. The optimum number would be equal to the number of processors (or processor cores) in the system.
Run-time checks. All the libraries are equipped with run-time checks to prevent bad parameters from crashing the software. You can disable this feature by setting PJ_ENABLE_EXTRA_CHECK to zero.
Stack checks. PJLIB is equipped with stack overflow detection. You can disable this feature by setting PJ_OS_HAS_CHECK_STACK to zero.
Safe module. PJSIP is equipped with mutex protection to protect PJSIP modules from being unregistered while they are still being accessed by PJSIP. If you don't plan to add/remove modules dynamically during run-time, you can disable this protection by setting PJSIP_SAFE_MODULE to zero.
Unescape in place. By default, PJSIP will make a copy of escaped message sequence before unescaping it. You can configure PJSIP to unescape in place by setting PJSIP_UNESCAPE_IN_PLACE to one.
Hash tolower optimization. By setting PJ_HASH_USE_OWN_TOLOWER to one, the hash function will convert the key to lower case and calculate the hash value in one loop.
Release mode. Don't forget to set the appropriate compiler optimization flag, and disable assertion with -DNDEBUG.

How can I configure pjsip to serve thousands of calls?

There are few settings to tweak:

First apply the CPU reduction techniques above to maximize the performance.
By default, PJSIP is configured to handle only 16384 simultaneous SIP transactions and dialogs. You should enlarge this to your requirement, by setting both PJSIP_MAX_TSX_COUNT and PJSIP_MAX_DIALOG_COUNT to the appropriate values (for example, 640*1024-1).
If you plan to support large number of TCP/TLS connections, then increase PJ_IOQUEUE_MAX_HANDLES to some large number (the default is only 64).

Footprint

How can I reduce the footprint and heap usage of my application?

With the default settings, PJSIP is not really optimized for size (nor speed), since the main objective is to have sufficient information for troubleshooting. Taking Windows as reference platform, the executable size of Release build with Visual C++ 6 is as follows:

  pjsua_vc6.exe: file size=749,672 bytes

      241664 .data
      102400 .rdata
       24576 .reloc
      536576 .text

  (all numbers in decimal, bytes)

Note: with default settings, pjsua will include PJNATH/ICE implementation as well as many media goodies, which you may not need in your embedded application. Hence the executable size is a bit large for pjsua.

The heap usage with 2 connected calls using Speex wideband codec and AEC enabled (press dd in pjsua menu):

>>> dd
..
  Total    908508 of   1046304 (86 %) used!
..

To reduce the executable size, you can apply the following settings:

Reduce logging verbosity. The default logging level is 5, which provides verbose information just in case some debugging is needed. You can decrease the logging verbosity level to 3 so that only vital information is displayed, by setting PJ_LOG_MAX_LEVEL macro (in config_site.h as usual). This will reduce executable size by approximately 28 KB.
Turn off logging. Alternatively you can disable logging altogether, by setting PJ_LOG_MAX_LEVEL to 0. This will reduce executable size by another 28 KB.
Disable Speex AEC, by setting PJMEDIA_HAS_SPEEX_AEC=0, to reduce executable size by 32 KB.
Disable resampling, by setting PJMEDIA_HAS_LIBRESAMPLE=0, to reduce executable size by 45 KB.
Disable unused codecs. To leave with only G.711 codecs (pcma and pcmu), disable speex with PJMEDIA_HAS_SPEEX_CODEC=0, disable iLBC with PJMEDIA_HAS_ILBC_CODEC=0, disable GSM codec with PJMEDIA_HAS_GSM_CODEC=0, and disable L16 codecs with PJMEDIA_HAS_L16_CODEC=0. This will reduce executable size by approximately 114 KB.
Disable alaw/ulaw table. By default, a table based alaw/ulaw implementation is used. You can disable this by setting PJMEDIA_HAS_ALAW_ULAW_TABLE=0, which makes PJMEDIA to calculate the alaw/ulaw value rather than using the table. This will reduce executable size by approximately 28 KB.
Disable error string. All libraries keep the description of the error codes in some static variables. You can omit this error description by setting PJ_HAS_ERROR_STRING=0. When the error description is omitted, pj_strerror() will just print the error code rather than the error description. You can then look for the error description in libraries source codes, or by searching on PJSIP website. Omitting the error description will reduce executable size by approximately 20 KB.
Disable run-time checks. All the libraries are equipped with run-time checks to prevent bad parameters from crashing the software. You can disable this feature by setting PJ_ENABLE_EXTRA_CHECK to zero. This will reduce executable size by approximately 20 KB.
Disable stack checks. PJLIB is equipped with stack overflow detection. You can disable this feature by setting PJ_OS_HAS_CHECK_STACK to zero, to reduce executable size by approximately 4 KB.
Disable CRC32 table, by setting PJ_CRC32_HAS_TABLES=0, to reduce executable size by about 1 KB, only if you use ICE.
Use your own sound device abstraction, rather than PortAudio. If you are porting PJSIP to an embedded platform, you may need to create your own sound device abstraction. So supposing we don't use PortAudio and use the NULL sound device implementation (PJMEDIA_SOUND_IMPLEMENTATION=PJMEDIA_SOUND_NULL_SOUND, we will reduce executable size by approximately 49 KB.

With all above optimizations set, we now have pjsua size (still with ICE/PJNATH and all media goodies included in the executable):

  pjsua_vc6.exe: file size= 381,032 bytes

   184320 .data
     8192 .rdata
    16384 .reloc
   319488 .text

Using the same settings, if we take the executable size of simpleua.exe (this is a sample program to do simple call with audio, without conference bridge nor ICE/STUN):

  simpleua.exe: file size= 155,648 bytes

    28672 .data
     4096 .rdata
   139264 .text

At this point, the heap memory usage of pjsua with 2 calls has been reduced by about 100 KB:

>>> dd
..
 Total    793624 of    909024 (87 %) used!
..

We still want more heap usage reductions, of course!

Now, assuming that the product will only need to support, say, 8 calls, we can apply these settings to reduce heap memory usage:

Transaction/dialog/call count. Set the maximum number of concurrent transactions/dialogs/calls with

#   define PJSIP_MAX_TSX_COUNT 		31
#   define PJSIP_MAX_DIALOG_COUNT 	31
#   define PJSUA_MAX_CALLS		31

Optimize pool sizes. These settings not only will reduce heap memory usage, but will also prevent the libraries from allocating too many large memory blocks. With the default settings, most memory pools are configured to allocate memory in 4KB blocks, and some system like Symbian will have difficulties in providing these blocks to PJSIP. Use the following setting to reduce the memory block size used by memory pools, at the expense of more calls to system's memory allocators (new or malloc) to allocate memory:

#   define PJSIP_POOL_LEN_ENDPT		1000
#   define PJSIP_POOL_INC_ENDPT		1000
#   define PJSIP_POOL_RDATA_LEN		2000
#   define PJSIP_POOL_RDATA_INC		2000
#   define PJSIP_POOL_LEN_TDATA		2000
#   define PJSIP_POOL_INC_TDATA		512
#   define PJSIP_POOL_LEN_UA		2000
#   define PJSIP_POOL_INC_UA		1000
#   define PJSIP_POOL_TSX_LAYER_LEN	256
#   define PJSIP_POOL_TSX_LAYER_INC	256
#   define PJSIP_POOL_TSX_LEN		512
#   define PJSIP_POOL_TSX_INC		128
#   define PJMEDIA_SESSION_SIZE		1000
#   define PJMEDIA_SESSION_INC		1000

With these settings applied, heap memory usage will be reduced very significantly. Looking at heap memory usage of pjsua with two G.711 calls:

pjsua_vc6 --clock-rate 8000 --ec-tail 0 --max-calls 2 --no-tcp

>>> dd
..
  Total    120532 of    150344 (80 %) used!
..

So only about 150 KB for two calls, I think that should be affordable.

Windows Specific Questions

How can I use PJSIP in .NET applications?

For this, you will need to have ActiveX wrapper for PJSIP. Unfortunately, at this point we don't provide ActiveX projects for PJSIP, but you can use one of the following projects:

Sipek Phone project, by created Sasa Coh, is a project to build a SIP softphone using .NET, and it has an PJSIP ActiveX wrapper as part of the project.