@Pai Peng Have you looked at pjsua_dump.c ? On Wed, Mar 13, 2013 at 5:01 AM, Pai Peng <sipaipv6 at gmail.com> wrote: > Hello, > > > Thank you for reply. > > > If I am using RTCP for network checking, is there any callback function > used to update the RTCP result to UI? > > Thanks > > Pai > > Sent from my iPhone > > On 11.03.2013, at 18:34, <gaurav.srivastava2 at agnity.com> wrote: > > > Hi Pai, > > > > Enable the feature RTCP on both the clients you will get the desired > values from PJSIP. > > > > Thanks and Regards, > > Gaurav > > > > > > On Mon, 11 Mar 2013 17:58:30 +0100, Pai Peng wrote: > > > Hello, > > > > > > > > > My new question is if it is possible to use pjsip to test VoIP audio > > > quality(jitter, paket lost, etc.)? Is there any functions implemented > for > > > this purpose? (or I need to calulate the values by myself?) > > > > > > Or any other open source? > > > > > > I would like to test the network broadband speed for audio quality, > which > > > audio codec to use for under different conditions. > > > > > > > > > Thank you. > > > > > > > > > Regards, > > > > > > Pai > > _______________________________________________ > > 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 > -- Khoa Pham HCMC University of Science Faculty of Information Technology -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20130313/d4b59a89/attachment-0001.html>
