Hi Benny, I have a question about the codec selection process. Now the Speex/16000 has the highest priority (is set in the code like that) but when I run the application only G711 is used. I could verify this using the debug version and put a breakpoint in "spx_codec_encode" function the application never stops there. It seems that only G711 codec is used. Is this because of the SIP server? The media line 'm' from the SDP how is managed (or negotiated) by the pjsip module? I do my tests with a phone Nokia E61 which is connected with Wi-Fi. Can I ask you on what Nokia (or Symbian) phones the library was tested (may be E61 is not a good candidate)? Thank you, George. 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 _______________________________________________ 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/20090409/99c06b77/attachment-0001.html>
