> > Hi, > > > > I am trying to make a (decent) VoIP call using the following setup: > > - Asterix PBX (latest version) running on Trixbox > > - pjsua (command line tool) on a Windows XP host PC > > - Cellphone (O2) running Windows Mobile 5 and pjsua_wince application > > > > FYI PocketPJ is the new (and better) app sample for WM I was not aware of its existence. However, I've tried it and the symptoms are pretty much the same - codec PCMU, echo, "silence" noise (which seems to be generated by the echo suppressor) > > > > > - LAN is connected using a LinkSys WRT54L (PC connected with cable and > > phone connected with WiFi) > > > > > First thing first, which pjsip version are you using on both sides of the > call? latest svn trunk > > > I manage to make a phone between the O2 and the PC, however the quality is > > very poor, with echo and disruptions of the voice. > > > > This probably is a known problem. We use echo suppressor on WM and it's > performance is quite poor. If anyone can suggest better alternatives then > that will be great. Yeah, I've noticed that the echo suppressor is quite poor. Will speex make things better, or it's a completely different issue? > > > Also, I periodically (~1 sec interval) get "silence noises" when there is > > no talk. > > > > > I suppose that's the VAD and keep-alive in action. This is also a known > issue: http://trac.pjsip.org/repos/ticket/490 > > > > dq dump from pjsua: > > [CONFIRMED] To: "801" <sip:801 at 10.0.0.151 <sip%3A801 at 10.0.0.151> > > >;tag=as18eb6d76 > > Call time: 00h:01m:12s, 1st res in 0 ms, conn in 0ms > > SRTP status: Not active Crypto-suite: (null) > > #0 PCMU @8KHz, sendrecv, peer=10.0.0.151:10406 > > RX pt=0, stat last update: 00h:00m:02.781s ago > > total 3.5Kpkt 570.8KB (713.6KB +IP hdr) @avg=63.1Kbps/78.8Kbps > > 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 3.517 21.375 4.000 2.393 > > TX pt=0, ptime=20ms, stat last update: 00h:00m:02.375s ago > > total 3.5Kpkt 575.4KB (719.3KB +IP hdr) @avg 63.6Kbps/79.5Kbps > > 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 : 18.875 21.696 24.375 22.375 1.661 > > RTT msec : 2.014 5.922 11.016 5.020 2.840 > > > > > > I have the following questions: > > 1. Why is the PCMU codec selected and not Speex (I have set Speex to be in > > the highest priority on both sides)? How can I make it choose something else > > other than PCMU? > > When I tries setting all Codecs priorities except Speex to zero, I couldn't > > establish a conversation. > > > > > Did you put/enable Speex in the build? > on the WM side I didn't change anything this round - so I guess it is included there by default. on the Desktop side speex is there (seeing it by typing Cp) Is there another way to confirm they both have it? do I need to specify something in the PocketPJ build? > > 2. I set --no-vad in the pjsua console on the PC and set: > > # define PJMEDIA_CODEC_MAX_SILENCE_PERIOD -1 > > before deploying the file to the mobile device. > > This should have disabled the 'silence noises' but it doesn't. Am I missing > > something? > > > > > Disabling the vad with --no-vad should be sufficient. Of course you'd need > to do the same in the WM side as well. > > done that...has no effect. > > > 3. Although I use non-routable IPs, I have the entire system on a LAN. Why > > does it think I'm behind a NAT and tunnels everything through the Asterix? > > Is there a way to tell the software to contact directly? > > I've set the following in pjsua_wince.cpp: > > // Use this to disable STUN > > # define STUN_DOMAIN NULL > > # define STUN_SERVER NULL > > And it does not seem to have any effect. > > > > > If you're calling an Asterisk number (or otherwise calling via Asterisk), > then it's up to your Asterisk configuration whether to tunnel the audio or > not, no? > I thought so too. I guess that since the entire network is behind NAT and uses non-routable IPs it thinks that it should tunnel everything through the Asterix. > > > 4. Was anyone able to succesfully deploy such (or similar) enviorment? Can > > you provide some hints? > > > > 5. It seems that the changes I make have very little or no effect on the > > overall quality. For example, I set: > > # define PJMEDIA_SOUND_BUFFER_COUNT 20 > > and expected to have a very long delay, but it had no effect. > > Am I doing something wrong? > > > > > With the latest (0.9) pjsip, that setting has less effect now as the buffer > count will be calculated dynamically. I see. > > Cheers > Benny
