It is 2.x release active now and the problem with "ALSA native support"?in pjsua application?still remains. While auddemo application (compiled with ALSA native support) ?works great both recording and playing sound through my aliased "duplex" alsa device (see asound.conf below) simultaneosly with other application Pjsua with same compilation option DOES NOT WORK, breaking with: >>> pjsua8: ../src/pjmedia/conference.c:1795: get_frame: Assertion `frame->size == conf->samples_per_frame * conf->bits_per_sample / 8' failed. It is worth noting that pjsua built with portaudio DOES WORK, BUT works unstable and sound stops playing after 10-30 seconds of playback! Using latest external libportaudio works a little bit more stable and sound plays a little bit longer and then also stops. A guy called Damir also had that problem and it seems that he could fix things up by asound.conf magic. That solution doesn't work for my hardware. I've?played a lot with?asoundrc options and aliases. Nothing helps. here's Damirs post: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2011-February/012591.html Some other guys recommend me using external sound mixing server, which doesn't suite my task very well: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2010-April/010721.html Problem exits. Noone yet had offered a fix, only workarounds. Maybe guys that created alsa_dev.c can shed some light on the problem? I really need help with this :( The only thing that works with ALSA and shared soundcard? is still pjsua version 1.0.3. It also trends to just stop playing sound but really really rare (twice a week). == Here's how pjsua crashes after local echotest (cc 0 0): == TC421:~# ./pjsua8 20:22:01.147 os_core_unix.c !pjlib 2.0.0-beta for POSIX initialized 20:22:01.148 sip_endpoint.c? .Creating endpoint instance... 20:22:01.151????????? pjlib? .select() I/O Queue created (0x81bae18) 20:22:01.151 sip_endpoint.c? .Module "mod-msg-print" registered 20:22:01.151 sip_transport.? .Transport manager created. 20:22:01.151?? pjsua_core.c? .PJSUA state changed: NULL --> CREATED 20:22:01.151 sip_endpoint.c? .Module "mod-pjsua-log" registered 20:22:01.151 sip_endpoint.c? .Module "mod-tsx-layer" registered 20:22:01.151 sip_endpoint.c? .Module "mod-stateful-util" registered 20:22:01.151 sip_endpoint.c? .Module "mod-ua" registered 20:22:01.151 sip_endpoint.c? .Module "mod-100rel" registered 20:22:01.151 sip_endpoint.c? .Module "mod-pjsua" registered 20:22:01.151 sip_endpoint.c? .Module "mod-invite" registered 20:22:01.213???? alsa_dev.c? ..ALSA driver found 8 devices 20:22:01.213???? alsa_dev.c? ..ALSA initialized 20:22:01.213????????? pjlib? ..select() I/O Queue created (0x81e94bc) 20:22:01.251 sip_endpoint.c? .Module "mod-evsub" registered 20:22:01.251 sip_endpoint.c? .Module "mod-presence" registered 20:22:01.251 sip_endpoint.c? .Module "mod-mwi" registered 20:22:01.251 sip_endpoint.c? .Module "mod-refer" registered 20:22:01.252 sip_endpoint.c? .Module "mod-pjsua-pres" registered 20:22:01.252 sip_endpoint.c? .Module "mod-pjsua-im" registered 20:22:01.252 sip_endpoint.c? .Module "mod-pjsua-options" registered 20:22:01.252?? pjsua_core.c? .1 SIP worker threads created 20:22:01.252?? pjsua_core.c? .pjsua version 2.0.0-beta for Linux-2.6.32.9/i686/glibc-2.9 initialized 20:22:01.252?? pjsua_core.c? .PJSUA state changed: CREATED --> INIT 20:22:01.252 sip_endpoint.c? Module "mod-default-handler" registered 20:22:01.252?? pjsua_core.c? SIP UDP socket reachable at 192.168.4.190:5060 20:22:01.252?? udp0x81f09a0? SIP UDP transport started, published address is 192.168.4.190:5060 20:22:01.252??? pjsua_acc.c? Adding account: id=<sip:192.168.4.190:5060> 20:22:01.252??? pjsua_acc.c? .Account <sip:192.168.4.190:5060> added with id 0 20:22:01.252??? pjsua_acc.c? Acc 0: setting online status to 1.. 20:22:01.253??? tcplis:5060? SIP TCP listener ready for incoming connections at 192.168.4.190:5060 20:22:01.253??? pjsua_acc.c? Adding account: id=<sip:192.168.4.190:5060;transport=TCP> 20:22:01.253??? pjsua_acc.c? .Account <sip:192.168.4.190:5060;transport=TCP> added with id 1 20:22:01.253??? pjsua_acc.c? Acc 1: setting online status to 1.. 20:22:01.253?? pjsua_core.c? PJSUA state changed: INIT --> STARTING 20:22:01.253? pjsua_media.c? ..NAT type detection failed: Invalid STUN server or server not configured (PJNATH_ESTUNINSERVER) 20:22:01.253 sip_endpoint.c? .Module "mod-unsolicited-mwi" registered 20:22:01.253?? pjsua_core.c? .PJSUA state changed: STARTING --> RUNNING >>>> Account list: ? [ 0] <sip:192.168.4.190:5060>: does not register ?????? Online status: Online ?*[ 1] <sip:192.168.4.190:5060;transport=TCP>: does not register ?????? Online status: Online Buddy list: ?-none- +=============================================================================+ |?????? Call Commands:???????? |?? Buddy, IM & Presence:? |???? Account:????? | |????????????????????????????? |????????????????????????? |?????????????????? | |? m? Make new call??????????? | +b? Add new buddy?????? .| +a? Add new accnt | |? M? Make multiple calls????? | -b? Delete buddy???????? | -a? Delete accnt. | |? a? Answer call????????????? |? i? Send IM????????????? | !a? Modify accnt. | |? h? Hangup call? (ha=all)??? |? s? Subscribe presence?? | rr? (Re-)register | |? H? Hold call??????????????? |? u? Unsubscribe presence | ru? Unregister??? | |? v? re-inVite (release hold) |? t? ToGgle Online status |? >? Cycle next ac.| |? U? send UPDATE????????????? |? T? Set online status??? |? <? Cycle prev ac.| | ],[ Select next/prev call??? +--------------------------+-------------------+ |? x? Xfer call??????????????? |????? Media Commands:???? |? Status & Config: | |? X? Xfer with Replaces?????? |????????????????????????? |?????????????????? | |? #? Send RFC 2833 DTMF?????? | cl? List ports?????????? |? d? Dump status?? | |? *? Send DTMF with INFO????? | cc? Connect port???????? | dd? Dump detailed | | dq? Dump curr. call quality? | cd? Disconnect port????? | dc? Dump config?? | |????????????????????????????? |? V? Adjust audio Volume? |? f? Save config?? | |? S? Send arbitrary REQUEST?? | Cp? Codec priorities???? |?????????????????? | +-----------------------------------------------------------------------------+ |? q? QUIT?? L? ReLoad?? sleep MS?? echo [0|1|txt]???? n: detect NAT type???? | +=============================================================================+ You have 0 active call >>> cc 0 0 20:22:08.552? pjsua_media.c? Conf connect: 0 --> 0 20:22:08.552? pjsua_media.c? .Set sound device: capture=-1, playback=-2 20:22:08.552??? pjsua_app.c? ..Turning sound device ON 20:22:08.552? pjsua_media.c? ..Opening sound device PCM at 16000/1/20ms 20:22:08.565??? ec0x81c4070? ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=99 ms 20:22:08.565?? conference.c? .Port 0 (duplex) transmitting to port 0 (duplex) Success >>> pjsua8: ../src/pjmedia/conference.c:1795: get_frame: Assertion `frame->size == conf->samples_per_frame * conf->bits_per_sample / 8' failed. == my asound.conf == pcm.dmixer? { ??????? type dmix ??????? ipc_key 1024 ??????? slave { ??????????? pcm "hw:0,0" ??????????? period_time 0 ??????????? rate 44100 ??????????? period_size 1024 ??????????? buffer_size 8192 ??????? } ??????? bindings { ??????????? 0 0 ??????????? 1 1 ??????? } ??? } pcm.conv_dmixer { ??????? type plug ??????? slave.pcm "dmixer" } pcm.conv_record { ??????? type plug ??????? slave.pcm "hw:0,0" } pcm.duplex { ?? type asym ?? playback.pcm "conv_dmixer" ?? capture.pcm "conv_record" ?? hint { ??????? show on ??????? description "Duplex device" ?? } } pcm.!default { ?? type plug ?? slave.pcm "duplex" } -------------- next part -------------- An HTML attachment was scrubbed... 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