RE : high RTP packet loss rate when using GSM over GPRS

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



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?









[Index of Archives]     [Asterisk Users]     [Asterisk App Development]     [Linux ARM Kernel]     [Linux ARM]     [Linux Omap]     [Fedora ARM]     [IETF Annouce]     [Security]     [Bugtraq]     [Linux]     [Linux OMAP]     [Linux MIPS]     [Linux API]
  Powered by Linux