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?
