Linux/PROCESSOR-SDK-OMAPL138: underrun/overrun issue while using codecs(speex16000, speex8000 , speex32000 ,GSM ,g722,pcma,pcmu)

Hi Yordan,

I am using TMDSLCDK138 development board.

I am using an opensource application pjsip which downloaded from "  "

procedure for build  it for linux x86 is following,

./configure

make dep

make

you can find the executable ~/pjsip-app/bin

To do cross compilation you need to export the toolchain path and add CC=(toolchain path)/arm-linux-gnueabi-gcc . then make dep and make.

Regards,

Allwyn

Hi Yordan,

I did the following steps to build it for arm,

in terminal,

export ARCH=arm

export PATH=~(path)/ti-processor-sdk-linux-omapl138-lcdk-04.02.00.09/linux-devkit/sysroots/x86_64-arago-linux/usr/bin/:$PATH

export CROSS_COMPILE=~(path)/ti-processor-sdk-linux-omapl138-lcdk-04.02.00.09/linux-devkit/sysroots/x86_64-arago-linux/usr/bin/arm-linux-gnueabi-

cd projectpath

./configure --host=arm-linux-gnueabi CC=(path)/ti-processor-sdk-linux-omapl138-lcdk-04.02.00.09/linux-devkit/sysroots/x86_64-arago-linux/usr/bin/arm-linux-gnueabi-gcc --disable-libwebrtc

make dep

make

then you will find the executable in ~pjsip-apps/bin 

run the executable in board.(pjsua-arm-unknown-linux-gnueabi).

you can disable codecs in a config file 

create a config file in the location of executable.

vi codec.cfg

in that file you can disable the codecs by doing the following("#" is used to comment)

--dis-codec=speex/16000
--dis-codec=speex/8000
--dis-codec=speex/32000
--dis-codec=iLBC/8000
--dis-codec=GSM/8000
--dis-codec=PCMU/8000
--dis-codec=PCMA/8000
#--dis-codec=G722/16000
--dis-codec=L16/44100

# to change the clock rate include the following in the file (remove the "#") /default is 16000

#--clock-rate=8000

To run the executable with following config file,

./pjsua-arm-unknown-linux-gnueabi --config-file codec.cfg

the PCMA/PCMU worked for me in 8000 clock rate and disabling all codecs.

regards

Allwyn

Hi Yordan,

I checked with new SDK on TMDSLCDK138. It shows the same error. Have you tried to make call to another application using pjsip.(ip call).

root@omapl138-lcdk:~/test# ./pjsua-arm-unknown-linux-gnueabi --clock-rate=8000
06:30:10.517 os_core_unix.c !pjlib 2.7.1 for POSIX initialized
06:30:22.554 sip_endpoint.c .Creating endpoint instance...
06:30:22.559 pjlib .select() I/O Queue created (0x1c5508)
06:30:22.560 sip_endpoint.c .Module "mod-msg-print" registered
06:30:22.560 sip_transport. .Transport manager created.
06:30:22.560 pjsua_core.c .PJSUA state changed: NULL --> CREATED
06:30:22.563 sip_endpoint.c .Module "mod-pjsua-log" registered
06:30:22.563 sip_endpoint.c .Module "mod-tsx-layer" registered
06:30:22.563 sip_endpoint.c .Module "mod-stateful-util" registered
06:30:22.564 sip_endpoint.c .Module "mod-ua" registered
06:30:22.565 sip_endpoint.c .Module "mod-100rel" registered
06:30:22.565 sip_endpoint.c .Module "mod-pjsua" registered
06:30:22.567 sip_endpoint.c .Module "mod-invite" registered
06:30:22.962 alsa_dev.c ..ALSA driver found 2 devices
06:30:22.963 alsa_dev.c ..ALSA initialized
06:30:22.965 pjlib ..select() I/O Queue created (0x1e4a7c)
06:30:23.001 sip_endpoint.c .Module "mod-evsub" registered
06:30:23.002 sip_endpoint.c .Module "mod-presence" registered
06:30:23.002 sip_endpoint.c .Module "mod-mwi" registered
06:30:23.003 sip_endpoint.c .Module "mod-refer" registered
06:30:23.003 sip_endpoint.c .Module "mod-pjsua-pres" registered
06:30:23.004 sip_endpoint.c .Module "mod-pjsua-im" registered
06:30:23.004 sip_endpoint.c .Module "mod-pjsua-options" registered
06:30:23.006 pjsua_core.c .1 SIP worker threads created
06:30:23.006 pjsua_core.c .pjsua version 2.7.1 for Linux-4.14.40/armv5tejl/glibc-2.25 initialized
06:30:23.006 pjsua_core.c .PJSUA state changed: CREATED --> INIT
06:30:23.006 sip_endpoint.c Module "mod-default-handler" registered
06:30:35.024 pjsua_core.c SIP UDP socket reachable at 192.168.2.95:5060
06:30:35.025 udp0x1d87d0 SIP UDP transport started, published address is 192.168.2.95:5060
06:30:35.025 pjsua_acc.c Adding account: id=<sip:192.168.2.95:5060>
06:30:35.026 pjsua_acc.c .Account <sip:192.168.2.95:5060> added with id 0
06:30:35.026 pjsua_acc.c Modifying account 0
06:30:35.027 pjsua_acc.c Acc 0: setting online status to 1..
06:30:47.047 tcptp:5060 SIP TCP listener ready for incoming connections at 192.168.2.95:5060
06:30:47.047 pjsua_acc.c Adding account: id=<sip:192.168.2.95:5060;transport=TCP>
06:30:47.048 pjsua_acc.c .Account <sip:192.168.2.95:5060;transport=TCP> added with id 1
06:30:47.048 pjsua_acc.c Modifying account 1
06:30:47.048 pjsua_acc.c Acc 1: setting online status to 1..
06:30:47.049 pjsua_core.c PJSUA state changed: INIT --> STARTING
06:30:47.049 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
06:30:47.049 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
06:30:47.050 main.c Ready: Success
>>>>
Account list:
[ 0] <sip:192.168.2.95:5060>: does not register
Online status: Online
*[ 1] <sip:192.168.2.95:5060;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
>>> m
(You currently have 0 calls)
Buddy list:
-none-

Choices:
0 For current dialog.
-1 All 0 buddies in buddy list
[1 - 0] Select from buddy list
URL An URL
<Enter> Empty input (or 'q') to cancel
Make call: sip:192.168.2.14
06:30:59.317 pjsua_call.c Making call with acc #1 to sip:192.168.2.14
06:30:59.317 pjsua_aud.c .Set sound device: capture=-1, playback=-2
06:30:59.318 pjsua_app.c ..Turning sound device -1 -2 ON
06:30:59.318 pjsua_aud.c ..Opening sound device (speaker + mic) PCM@8000/1/20ms
06:30:59.458 ec0x1e3098 ...AEC created, clock_rate=8000, channel=1, samples per frame=160, tail length=200 ms, latency=0 ms
06:30:59.465 tcpc0x20c5c4 .TCP client transport created
06:30:59.479 tcptp:5060 !TCP listener 192.168.2.95:5060: got incoming TCP connection from 192.168.2.95:44777, sock=8
06:30:59.481 tcpc0x20c5c4 !.TCP transport 192.168.2.95:44777 is connecting to 192.168.2.95:5060...
06:30:59.481 tcps0xb4e0068c !TCP server transport created
06:30:59.483 pjsua_app.c SIP TCP transport is connected to [192.168.2.95:44777]
06:30:59.483 tcpc0x20c5c4 TCP transport 192.168.2.95:44777 is connected to 192.168.2.95:5060
06:30:59.484 pjsua_app.c SIP TCP transport is connected to [192.168.2.95:5060]
06:30:59.485 pjsua_media.c .Call 0: initializing media..
06:31:00.669 alsa_dev.c pb_thread_func: underrun!
06:31:03.749 alsa_dev.c !pb_thread_func: underrun!
06:31:04.304 sound_port.c EC suspended because of inactivity
06:31:11.505 pjsua_media.c !..RTP socket reachable at 192.168.2.95:4000
06:31:11.505 pjsua_media.c ..RTCP socket reachable at 192.168.2.95:4001
06:31:11.506 pjsua_media.c ..Media index 0 selected for audio call 0
06:31:11.515 pjsua_core.c ....TX 1111 bytes Request msg INVITE/cseq=17570 (tdta0x213024) to UDP 192.168.2.14:5060:
INVITE sip:192.168.2.14 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.95:44777;rport;branch=z9hG4bKPjd6ef7731-fd7f-4c93-ac37-614f2ca00c93
Max-Forwards: 70
From: <sip:192.168.2.95>;tag=a9580950-411f-4a0e-9931-2054a35d6dd4
To: sip:192.168.2.14
Contact: <sip:192.168.2.95:5060;ob>
Call-ID: ddbc9db6-1a47-4ba9-bc95-6f318af2ff5d
CSeq: 17570 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v2.7.1 Linux-4.14.40/armv5tejl/glibc-2.25
Content-Type: application/sdp
Content-Length: 473

v=0
o=- 3745895471 3745895471 IN IP4 192.168.2.95
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.2.95
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.2.95
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

--end msg--
06:31:11.517 pjsua_app.c .......Call 0 state changed to CALLING
>>> 06:31:11.519 pjsua_core.c .RX 323 bytes Response msg 100/INVITE/cseq=17570 (rdata0x1dd55c) from UDP 192.168.2.14:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.95:44777;rport=5060;received=192.168.2.95;branch=z9hG4bKPjd6ef7731-fd7f-4c93-ac37-614f2ca00c93
Call-ID: ddbc9db6-1a47-4ba9-bc95-6f318af2ff5d
From: <sip:192.168.2.95>;tag=a9580950-411f-4a0e-9931-2054a35d6dd4
To: <sip:192.168.2.14>
CSeq: 17570 INVITE
Content-Length: 0

--end msg--
06:31:18.343 pjsua_core.c .RX 898 bytes Response msg 200/INVITE/cseq=17570 (rdata0xb4e040bc) from UDP 192.168.2.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.95:44777;rport=5060;received=192.168.2.95;branch=z9hG4bKPjd6ef7731-fd7f-4c93-ac37-614f2ca00c93
Call-ID: ddbc9db6-1a47-4ba9-bc95-6f318af2ff5d
From: <sip:192.168.2.95>;tag=a9580950-411f-4a0e-9931-2054a35d6dd4
To: <sip:192.168.2.14>;tag=Do.TMddUQE5YDHTOV4I0cOmogKZYvzEJ
CSeq: 17570 INVITE
Contact: <sip:192.168.2.14:5060>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 275

v=0
o=- 3745895773 3745895774 IN IP4 192.168.2.14
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 98 96
c=IN IP4 192.168.2.14
b=TIAS:64000
a=rtcp:4009 IN IP4 192.168.2.14
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16

--end msg--
06:31:18.344 pjsua_app.c .....Call 0 state changed to CONNECTING
06:31:18.346 pjsua_media.c .....Call 0: updating media..
06:31:18.347 pjsua_media.c .......Media stream call00:0 is destroyed
06:31:18.347 pjsua_aud.c ......Audio channel update..
06:31:18.351 strm0xb4e0723c .......VAD temporarily disabled
06:31:18.353 strm0xb4e0723c .......Encoder stream started
06:31:18.353 strm0xb4e0723c .......Decoder stream started
06:31:18.355 pjsua_media.c ......Audio updated, stream #0: speex (sendrecv)
06:31:18.355 pjsua_app.c .....Call 0 media 0 [type=audio], status is Active
06:31:18.355 pjsua_aud.c .....Conf connect: 3 --> 0
06:31:18.356 conference.c ......Port 3 (sip:192.168.2.14) transmitting to port 0 (default:CARD=LCDK)
06:31:18.356 pjsua_aud.c .....Conf connect: 0 --> 3
06:31:18.356 conference.c ......Port 0 (default:CARD=LCDK) transmitting to port 3 (sip:192.168.2.14)
06:31:18.359 pjsua_core.c .....TX 361 bytes Request msg ACK/cseq=17570 (tdta0xb4e19b0c) to UDP 192.168.2.14:5060:
ACK sip:192.168.2.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.95:44777;rport;branch=z9hG4bKPjd6f30b0c-140f-4216-9707-764d41638b8e
Max-Forwards: 70
From: <sip:192.168.2.95>;tag=a9580950-411f-4a0e-9931-2054a35d6dd4
To: sip:192.168.2.14;tag=Do.TMddUQE5YDHTOV4I0cOmogKZYvzEJ
Call-ID: ddbc9db6-1a47-4ba9-bc95-6f318af2ff5d
CSeq: 17570 ACK
Content-Length: 0

--end msg--
06:31:18.360 pjsua_app.c .....Call 0 state changed to CONFIRMED
06:31:18.482 sound_port.c EC activated
06:31:19.036 alsa_dev.c pb_thread_func: underrun!
06:31:19.074 Master/sound Buffer size adjusted from 1018 to 859 (eff_cnt=640)
06:31:19.656 Master/sound Buffer size adjusted from 1014 to 877 (eff_cnt=640)
06:31:20.111 alsa_dev.c pb_thread_func: underrun!
06:31:20.156 Master/sound Buffer size adjusted from 1037 to 954 (eff_cnt=640)
06:31:20.656 Master/sound Buffer size adjusted from 1075 to 890 (eff_cnt=640)
06:31:21.116 alsa_dev.c pb_thread_func: underrun!
06:31:21.157 Master/sound Buffer size adjusted from 987 to 888 (eff_cnt=640)
06:31:21.657 Master/sound Buffer size adjusted from 1075 to 838 (eff_cnt=640)
06:31:22.194 alsa_dev.c pb_thread_func: underrun!
06:31:23.259 alsa_dev.c pb_thread_func: underrun!
06:31:24.306 alsa_dev.c pb_thread_func: underrun!
06:31:25.447 alsa_dev.c pb_thread_func: underrun!
06:31:26.453 pjsua_core.c .RX 421 bytes Request msg BYE/cseq=28373 (rdata0xb4e040bc) from UDP 192.168.2.14:5060:
BYE sip:192.168.2.95:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.2.14:5060;rport;branch=z9hG4bKPjg.mqPkv0DYF.-G5mTWvQe0YeSCAe2JvJ
Max-Forwards: 70
From: <sip:192.168.2.14>;tag=Do.TMddUQE5YDHTOV4I0cOmogKZYvzEJ
To: <sip:192.168.2.95>;tag=a9580950-411f-4a0e-9931-2054a35d6dd4
Call-ID: ddbc9db6-1a47-4ba9-bc95-6f318af2ff5d
CSeq: 28373 BYE
User-Agent: PJSUA v2.7.1 Linux-4.15.0.34/x86_64/glibc-2.23
Content-Length: 0

--end msg--
06:31:26.456 pjsua_core.c .......TX 348 bytes Response msg 200/BYE/cseq=28373 (tdta0xb4e1d46c) to UDP 192.168.2.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.14:5060;rport=5060;received=192.168.2.14;branch=z9hG4bKPjg.mqPkv0DYF.-G5mTWvQe0YeSCAe2JvJ
Call-ID: ddbc9db6-1a47-4ba9-bc95-6f318af2ff5d
From: <sip:192.168.2.14>;tag=Do.TMddUQE5YDHTOV4I0cOmogKZYvzEJ
To: <sip:192.168.2.95>;tag=a9580950-411f-4a0e-9931-2054a35d6dd4
CSeq: 28373 BYE
Content-Length: 0

--end msg--
06:31:26.457 pjsua_app.c !......Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
06:31:26.546 pjsua_app_comm ......
[DISCONNCTD] To: sip:192.168.2.14;tag=Do.TMddUQE5YDHTOV4I0cOmogKZYvzEJ
Call time: 00h:00m:08s, 1st res in 18884 ms, conn in 18899ms
#0 audio speex @16kHz, sendrecv, peer=192.168.2.14:4008
SRTP status: Not active Crypto-suite:
RX pt=98, last update:00h:00m:07.954s ago
total 101pkt 5.6KB (9.6KB +IP hdr) @avg=5.4Kbps/9.4Kbps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 1.187 24.044 39.687 28.562 8.186
TX pt=98, ptime=20, last update:00h:00m:00.088s ago
total 15pkt 1.0KB (1.6KB +IP hdr) @avg=1.0Kbps/1.6Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 1.129 1.384 1.724 1.724 0.245
06:31:26.547 pjsua_media.c ......Call 0: deinitializing media..
06:31:26.558 pjsua_media.c ........Media stream call00:0 is destroyed
06:31:26.560 alsa_dev.c pb_thread_func: underrun!
06:31:26.724 alsa_dev.c !pb_thread_func: underrun!
06:31:27.044 alsa_dev.c !pb_thread_func: underrun!
06:31:27.626 pjsua_aud.c !Closing sound device after idle for 1 second(s)
06:31:27.627 pjsua_app.c .Turning sound device -1 -2 OFF
06:31:27.627 pjsua_aud.c .Closing default:CARD=LCDK sound playback device and default:CARD=LCDK sound capture device
06:31:32.486 tcpc0x20c5c4 TCP transport destroyed normally
06:31:32.487 tcps0xb4e0068c TCP connection closed
06:31:32.487 pjsua_app.c SIP TCP transport is disconnected from [192.168.2.95:44777]: End of file (PJ_EEOF) [status=70016]
06:31:32.487 pjsua_acc.c Disconnected notification for transport tcps0xb4e0068c
06:31:32.488 sip_transport. .Transport tcps0xb4e0068c shutting down, force=0
06:31:32.489 tcps0xb4e0068c TCP transport destroyed with reason 70016: End of file (PJ_EEOF)

regards,

Allwyn