Hi, can anyone tell me the number of packet per second sent using gsm codec, and g711 codec. And what is the exact inter frame time? Ex: gsm codec is a 8Khz codec, and has a default ptime set to 20 ms; and packet size is 33 bytes. right? does this mean that gsm codec buffer receives (8 * 20 )bits per frame???? (so 20 bytes)? how are these bytes turned into UDP RTP packets? g711 is 64khz codec. if ptime is 20ms => 1 frame = 64 * 20 /8 = 160 bytes ??? How much RTP packets are generated here??? Any help ??? -------- Message d'origine-------- De: pjsip-bounces at lists.pjsip.org de la part de Olle Frimanson Date: jeu. 30/06/2011 14:32 ?: 'pjsip list' Objet : Re: [pjsip]RE?: RE?: RE?: high RTP packet loss rate when using GSM over GPRS You could have a look at OpenCore AMR. I don't know about the GSM , to high bitrate for the scenario you are targeting I think. BR/Olle -----Ursprungligt meddelande----- Fr?n: pjsip-bounces at lists.pjsip.org [mailto:pjsip-bounces at lists.pjsip.org] F?r Idy Thiam Skickat: den 30 juni 2011 14:29 Till: pjsip list ?mne: [pjsip] RE?: RE?: RE?: high RTP packet loss rate when using GSM over GPRS Hi, ping is OK, with a 40 to 450 ms delay. vad is enabled, I would like to adapt silence threshold, but I don't see how to do it, I tried default settings but also changed threshold like this: status = pjmedia_silence_det_set_adaptive(gsm_data->vad,1000); if (status != PJ_SUCCESS) { PJ_LOG(1,(__FILE__,"Error %d when setting silence threshold.",status)); status = PJ_SUCCESS; } else { PJ_LOG(3,(__FILE__,"Silence detector thresold SET.")); } status = pjmedia_silence_det_set_params(gsm_data->vad,4000,5000,1000); if (status != PJ_SUCCESS) { PJ_LOG(1,(__FILE__,"Error %d when setting silence parameters.",status)); status = PJ_SUCCESS; } AMR-NB is not possible here since I run an ARM based architecture, not x86. I tried a 8 frame per packet configuration with gsm codec, no success... What is the flow with gsm codec? how much RTP packets are sent per second? Is it possible to control this rate? Thanks for your help... -------- Message d'origine-------- De: pjsip-bounces at lists.pjsip.org de la part de Olle Frimanson Date: mer. 29/06/2011 15:09 ?: 'pjsip list' Objet : Re: [pjsip]RE?: RE?: high RTP packet loss rate when using GSM over GPRS Hi yes it should decrease, You could have a look at the jitterbuffer parameters and RTT if you do a dq in pjsua. Se if you have very long time. If you just ping over GPRS does all pings go through. VAd can help on the bandwidth as well. BR/Olle -----Ursprungligt meddelande----- Fr?n: pjsip-bounces at lists.pjsip.org [mailto:pjsip-bounces at lists.pjsip.org] F?r Idy Thiam Skickat: den 29 juni 2011 14:21 Till: pjsip list ?mne: [pjsip] RE?: RE?: high RTP packet loss rate when using GSM over GPRS Hi, I tried a 8 frame per packet config in the gsm codec, staying with a 20ms frame time, but I always have something like 80% RTP packet loss. Here are the results with various fr/pkt conf: 1frm/pkt (30kbps), 2 (21kbps), 4 (15kbps), 8(25kbps). The result for 8 frame per packet is no sense to me, since it should decrease. Right? Btw, I always loose 80% RTP packet. I am trying to set the frame time to 160, while setting the frame/packet to 8. Any suggestion will be helpful. -------- Message d'origine-------- De: pjsip-bounces at lists.pjsip.org de la part de Olle Frimanson Date: mar. 28/06/2011 16:32 ?: 'pjsip list' Objet : Re: [pjsip]RE?: high RTP packet loss rate when using GSM over GPRS AMR-NB is supported on x86 platforms bu using IPP codec's. You need to reduce the RTP/UDP/IP much more go to 8 frames/packet. This will give an overhead of 2kbps. But I don't think Asterisk supports this btw. BR/Olle -----Ursprungligt meddelande----- Fr?n: pjsip-bounces at lists.pjsip.org [mailto:pjsip-bounces at lists.pjsip.org] F?r Idy Thiam Skickat: den 28 juni 2011 15:08 Till: pjsip list ?mne: [pjsip] RE?: high RTP packet loss rate when using GSM over GPRS Hi, When using wireshark on the receiver side, I can see that more than 3000 rtp packets are sent by pjsua, within a 1 mn exchange. and the loss is 80%. I enabled vad, but it seems like this has no effect. I use asterisk for ip translation, but since I receive some packets, I think there is no side effects. I tried a 2 frame per packet in gsm codec, no success. Is AMR-NB supported by pjsip??? -------- Message d'origine-------- De: pjsip-bounces at lists.pjsip.org de la part de Olle Frimanson Date: mar. 28/06/2011 14:15 ?: 'pjsip list' Objet : Re: [pjsip] high RTP packet loss rate when using GSM over GPRS Hi the problem is that if you send one voice frame (ptime=20 ms) the overhead for RTP/UDP/IP corresponds to 16 kbps. On top of that you have the codec bitrate so the total bitrate will be around 30 kbps. A GPSR link can be as low as 8 kbps. Try increasing ptime to 160 ms (will give more delay) and switch to a better codec like AMR-NB. Finally you have no QoS in GPRS so packet delay can be huge up to 10 s sometimes. BR/Olle Fr?n: pjsip-bounces at lists.pjsip.org [mailto:pjsip-bounces at lists.pjsip.org] F?r Idy Thiam Skickat: den 28 juni 2011 12:15 Till: pjsip at lists.pjsip.org ?mne: [pjsip] high RTP packet loss rate when using GSM over GPRS Hi All, I am trying to use pjsip with a GSM module to communicate with a remote server. When using ADSL, there is no packet loss but with GPRS link, I have something like 80% loss. I think this is due to the fact that RTP is encapsulated into UDP, and there is no reliability, nor integrity. I have a 8Khz sound chip, and use gsm codecs. I think adapting rtp flow could be helpful, but I don t know how to handle it. Any help? _______________________________________________ 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 _______________________________________________ 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 -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5563 bytes Desc: not available URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20110701/d73f6e83/attachment.bin>
