Same, every 20ms a packet with 160byte + RTP overhead (min 12 bytes IIRC). You may have a different seeting to have a 30ms perfiod, then each packet contains 240bytes + RTP overhead. AFAIK g711 does not support features to squeeze several periods in one large RTP packet. Best regards, Werner Am 01.07.2011 16:23, schrieb Idy Thiam: > 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 > > > > > _______________________________________________ > 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
