2009/4/2 George Evi <george.evi at ctcinc.ca> > 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: > > Great! We love logs and 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<sip%3A5148403000 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? > We get these values from the RTCP report sent by the remote peer. If remote peer doesn't support RTCP, we would not get these stats of course. > And also why in the same section the loss period and jitter buffer values > are the same for all statistics colons? > > > It's probably because it's only got one RTCP report? In this case then the min/avg/max values would be the same, isn't it? cheers Benny > Thanks, > > George. > > > ------------------------------ > > *From:* pjsip-bounces at lists.pjsip.org [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: [pjsip] 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 at lists.pjsip.org [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: [pjsip] 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 > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090403/ce0fcfa1/attachment-0001.html>
