Hi Benny,
I activate the log in my project (PJ_LOG()) and used the function
"log_call_dump()" to dump the statistics at the end of a call.
I got these statistics:
--end msg--State changed from Null to Calling, event=TX_MSGTransaction
tsx0x732e8c state changed to Calling
[CONFIRMED] To: sip:5148403000 at sip6.van.netvoice.ca;tag=as5b69601c
Call time: 00h:01m:56s, 1st res in 3542 ms, conn in 5824ms
SRTP status: Not active Crypto-suite: (null)
#0 PCMU @8KHz, sendrecv, peer=64.34.49.82:19122
RX pt=0, stat last update: 00h:00m:00.601s ago
total 5.7Kpkt 924.8KB (1.15MB +IP hdr) @avg=62.3Kbps/77.9Kbps
pkt loss=143 (2.4%), discrd=1 (0.0%), dup=0 (0.0%), reord=1 (0.0%)
(msec) min avg max last dev
loss period: 20.000 178.750 940.000 100.000 63.638
jitter : - 0.001 11.737 579.000 0.750 8.775
TX pt=0, ptime=20ms, stat last update: 00h:00m:03.680s ago
total 5.7Kpkt 906.4KB (1.13MB +IP hdr) @avg 61.1Kbps/76.5Kbps
pkt loss=2 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 40.000 40.000 40.000 40.000 0.000
jitter : 193.000 193.000 193.000 193.000 0.000
RTT msec : 92.000 128.637 332.000 101.000 21.526
Processing incoming message: Response msg 200/BYE/cseq=18255
(rdata0x71e65c)RX 510 bytes Response msg 200/BYE/cseq=18255 (rdata0x71e65c)
from UDP 64.34.49.82:5060:
SIP/2.0 200 OK
I also read the "Understanding Media Flow" document and I have a (beginner)
question.
In the TX section we have a jitter line but in the Media Flow diagram there
is no Jitter Buffer for packet transmission, what represents this line? And
also why in the same section the loss period and jitter buffer values are
the same for all statistics colons?
Thanks,
George.
_____
From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Wednesday, March 25, 2009 3:56 AM
To: pjsip list
Subject: Re: Pjlib on Symbian
2009/3/24 George Evi <george.evi at ctcinc.ca>
Hi Benny,
I update my application with the latest version the "trunk pjproject- 1.1"
and continue to test on Nokia E61.
As you suggested I changed the codec priorities in a way that GSM had
highest priority (in function "pjsua_media_subsys_init" priority value =
PJMEDIA_CODEC_PRIO_NORMAL +4 (132)). The sound was acceptable on the caller
side but on the callee side continue to be stuttered, disrupted and
instable.
Hi George,
In that case, I would probably suggest to try to use different peer for the
testing (pjsua running on desktop would be a good candidate :) ). And make
sure the audio doesn't get routed through the server or otherwise this
wouldn't make any difference. You can call directly to the device's IP
address to make sure.
Also I tried the others codecs (iLBC, Speex/8000 and Speex/16000) in the
same way but didn't see any improvements.
iLBC and Speex/16000 is definitely out of question. Speex/8000 is probably
bang on the processing capability, so use it with care (e.g. only use
release mode), and probably is not good for troubleshooting problems like
this. And of course there is G.711, definitely a good candidate to try.
cheers
Benny
Do you have any suggestions or ideas?
We don't want to use Nokia APS (Audio proxy Server) for the moment, because
it needs a publisher ID.
Thank you,
George.
_____
From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Friday, March 20, 2009 2:32 AM
To: pjsip list
Subject: Re: Pjlib on Symbian
2009/3/19 George Evi <george.evi at ctcinc.ca>
Hi Benny,
Thanks for your response.
The flow of sound is disrupted on both sides (caller and callee voice
reception). You can hear the sound but the words are not completed and on
the callee side the voice is metalique (like a robot speech). The latest
tests I made were done with Nokia E61 (S60 3rd edition -mr) connected Wi-Fi
and I expected to see some improvements but voice stilled disrupted.
I'm using iLBC as codec (1st priority) and UDP transport.
Aha, that's probably the reason. iLBC is heavy, I don't think the device has
enough processing power to run it [2]. Try with GSM or Speex.
Alternatively, consider using APS-Direct [1], available in pjsip version 1.1
now downloadable from the website. APS-Direct uses handset's native codec
and it supports iLBC, AMR, G.729, and G.711.
cheers
Benny
[1] http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct
[2] http://trac.pjsip.org/repos/wiki/PJMEDIA-MIPS
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