Hi Benny,
Thanks for your response. I have already tested in LAN and more than that on
Wi-Fi connection and it works fine.
Also I tested on wireless network (Access Point Rogers Canada) with PCMU and
we got a good voice quality.
We try to use a codec with less bandwidth (low bit rate) like GSM or iLBC.
Unfortunelly our Sip service provider supports for the moments only PCMU,
GSM-FR & iLBC mode 30.
I tried last week iLBC but for some reason after SIP session when media
session starts the application blocks (may be a bug ???). I'll continue to
investigate this problem.
Could you tell me, how did you test client/pjsua on E70 with different
codecs?
When I do such tests I set the priority at a higher level then the other
codecs (e.g. for GSM) in "pjsua_media_subsys_init" (pjsua_media.c) and I
comment the section of code in function "process_m_answer" (sdp_neg.c):
/* Arrange format in the offer so the order match the priority
* in the answer
*/
//!ge5may09 for (i=0; i<answer->desc.fmt_count; ++i) {
// unsigned j;
// pj_str_t *fmt = &answer->desc.fmt[i];
//
// for (j=i; j<offer->desc.fmt_count; ++j) {
// if (pj_strcmp(fmt, &offer->desc.fmt[j])==0) {
// str_swap(&offer->desc.fmt[i], &offer->desc.fmt[j]);
// break;
// }
// }
// }
, because my Voip provider sends always in this order (priority)
CSeq: 7436 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:5145294251 at 64.34.49.82>
Content-Type: application/sdp
Content-Length: 307
v=0
o=root 2719 2719 IN IP4 64.34.49.82
s=session
c=IN IP4 64.34.49.82
t=0 0
m=audio 18934 RTP/AVP 0 3 113 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:113 iLBC/8000
a=fmtp:113 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--end msg--Incoming Response msg 183/INVITE/cseq=7436 (rdata0x71e46c) in
state ProceedingState changed from Proceeding to Proceeding,
event=RX_MSGReceived
, always the PCMU in 1st position. Without comments the codec manager has
all the time this order of codecs.
Is there another way to have the same result?
Thanks for your support,
George.
_____
From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Tuesday, May 12, 2009 12:15 PM
To: pjsip list
Subject: Re: audio problem with GSM on Symbian
Hi George,
Thanks for the info. I think it's best to test in LAN first, direct to
another client/pjsua. I agree that it should work too with your setup, but
at this stage probably it's better to verify that the app works. And it
should be better to use the latest 1.2 release too, since we just tested
that last week (we tested with Speex/8000, PCMU, and G.722.1 on an older
E70).
cheers
Benny
On Tue, May 12, 2009 at 3:37 PM, George Evi <george.evi at ctcinc.ca> wrote:
Hi Benny,
I'm using MDA (no APS). The log file I usually open with MS Word may be that
the reason you don't have the expected results.
The test was on Nokia E71 connected to my wireless provider Access Point
(Rogers Canada) calling a PSTN local number.
Thanks,
George.
_____
From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Benny Prijono
Sent: Tuesday, May 12, 2009 10:08 AM
To: pjsip list
Subject: Re: audio problem with GSM on Symbian
Hi George,
First of all, are you using MDA or APS? Though we tested both before 1.2
release and they seem to be fine.
And secondly, to localize the problem, what if you call another pjsua on the
same LAN first and see how it goes.
cheers
Benny
PS: the log seems to strip most of the newlines so can't read too much info
there.
On Thu, May 7, 2009 at 7:43 PM, George Evi <george.evi at ctcinc.ca> wrote:
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
<http://trac.pjsip.org/repos/wiki/audio-problem-breakups> 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
<mailto:sip%3A5145294251 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
<mailto:sip%3A5145294251 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
<http://trac.pjsip.org/repos/wiki/audio-problem-dropouts> drop-outs or
"stutters":
1. Checking that Microphone and Speaker are
Functioning Properly, N/A on the phone.
2. Check that there is no dangling call in the PBX
<http://trac.pjsip.org/repos/wiki/audio-check-dangling-pbx-call> , N/A on
the phone.
3. Checking for Network Impairments of Incoming RTP
Packets, done in previous section.
4. Check
<http://trac.pjsip.org/repos/wiki/audio-check-cpu> that 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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