Hi,
I'm trying to use GSM/8000 codec with Pjsip library and have some audio
problems. Tests are done on a Nokia phone E71.
I passed through the following principal points (others which are not
present some of them are not applicable):
2. Audio breakups
<http://trac.pjsip.org/repos/wiki/audio-problem-breakups> :
1. It's always recommended to check whether the
problem exists with the latest SVN version of the libraries, Yes (revision
2668).
2. Check that CPU utilization is not too high, Yes
(can see a CPU load 19.47%, memory usage, function calls in profiler file).
3. Check for high network jitter, packet loss, Yes
(below is the statistical dump):
1. Pjsua Statistics, 5may09, 17h12:
1.1 GSM statistics ():
--end msg--State changed from Null to Calling, event=TX_MSGTransaction
tsx0x749e5c state changed to Calling
[CONFIRMED] To: sip:5145294251 at sip6.van.netvoice.ca;tag=as16e1281f
Call time: 00h:00m:59s, 1st res in 2211 ms, conn in 21845ms
SRTP status: Not active Crypto-suite: (null)
#0 GSM @8KHz, sendrecv, peer=64.34.49.82:14512
RX pt=3, stat last update: 00h:00m:00.525s ago
total 3.7Kpkt 125.3KB (277.1KB +IP hdr) @avg=12.5Kbps/27.8Kbps
pkt loss=171 (4.3%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 680.000 1,140.000 1,900.000 680.000 62.227
jitter : 0.250 12.555 4,135.000 0.250 8.533
TX pt=3, ptime=20ms, stat last update: 00h:00m:04.596s ago
total 3.1Kpkt 103.1KB (228.4KB +IP hdr) @avg 10.3Kbps/22.9Kbps
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 : 41.875 112.100 206.875 75.625 10.972
RTT msec : 152.000 223.800 540.000 152.000 17.061
Jitter buffer empty (prefetch=19)Starting talk burst (level=58
threshold=50)Start talksprut..Processing incoming message: Response msg
200/BYE/cseq=21287 (rdata0x71e46c)RX 507 bytes Response msg
200/BYE/cseq=21287 (rdata0x71e46c) from UDP 64.34.49.82:5060:
SIP/2.0 200 OK
- below are the statistics after using the PCMU codec:
1.2 PCMU statistics:
--end msg--State changed from Null to Calling, event=TX_MSGTransaction
tsx0x74a09c state changed to Calling
[CONFIRMED] To: sip:5145294251 at sip6.van.netvoice.ca;tag=as013a5bcb
Call time: 00h:00m:39s, 1st res in 2484 ms, conn in 24565ms
SRTP status: Not active Crypto-suite: (null)
#0 PCMU @8KHz, sendrecv, peer=64.34.49.82:18934
RX pt=0, stat last update: 00h:00m:03.806s ago
total 3.0Kpkt 491.8KB (614.8KB +IP hdr) @avg=63.7Kbps/79.6Kbps
pkt loss=0 (0.0%), discrd=1 (0.0%), dup=0 (0.0%), reord=1 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : - 0.001 6.563 598.000 11.250 6.433
TX pt=0, ptime=20ms, stat last update: 00h:00m:01.676s ago
total 2.6Kpkt 428.3KB (535.4KB +IP hdr) @avg 55.5Kbps/69.3Kbps
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 : 64.250 83.802 95.125 86.750 8.723
RTT msec : 173.000 209.417 283.000 197.000 64.301
Processing incoming message: Response msg 200/BYE/cseq=7437
(rdata0x71e46c)RX 506 bytes Response msg 200/BYE/cseq=7437 (rdata0x71e46c)
from UDP 64.34.49.82:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
4. Check if audio is breaking up when playing file locally, Yes (run
"sndtest.c").
3. Audio drop-outs
<http://trac.pjsip.org/repos/wiki/audio-problem-dropouts> or "stutters":
1. Checking that Microphone and Speaker are
Functioning Properly, N/A on the phone.
2. Check
<http://trac.pjsip.org/repos/wiki/audio-check-dangling-pbx-call> that there
is no dangling call in the PBX, N/A on the phone.
3. Checking for Network Impairments of Incoming RTP
Packets, done in previous section.
4. Check that CPU
<http://trac.pjsip.org/repos/wiki/audio-check-cpu> Utilization is not Too
High, already check in the previous section.
5. Try to enlarge PJMEDIA_SOUND_BUFFER_COUNT, no
need for the Pjsip version greater then 0.9.
6. Check for audio underflow/overflow, Yes (found jn the log file
traces of "underflow" situation, see attached "pjsua_MyE71_GSM_02.log" log
file).
The result of test:
- caller side: when listen welcome message voice is with no
breakups, no stuttering.
- callee (remote ) side: breakups, stuttering & echo present,
bad quality.
- I have also a WAV file example sent to Benny.
Thanks,
George.
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