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

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Hi, thanks for reply.

I tried pjsip in my laptop, with GPRS connexion, and I have the same amount of rtp loss (80%).
Using wireshark shows that the sending timing is not clear to me.

I tried g711 codec first, and gsm codec second.

g711: ptime =10ms, frame_per_pkt = 2 => 1 RTP packet should be sent each 20ms, with a 160 byte payload.

wireshark tells me that 1st pkt is after 30ms, and second is after 3?s => 2 pkts are sent each 30 ms (is this du to buffer???)

gsm: ptime = 20ms, clk = 8000 bps => 20ms = 20 bytes.

packet_size = 33 bytes => how does the codecs reach the 33 bytes, since after 20ms it only has 20bytes? delay???

I tried, in gsm_codec_parse(), a 20 bytes payload, and 2 or 3 packets are sent each 20 ms, instead of 1.
what does it mean?

I just want to make sure 1 pkt is sent each 10 or 20 ms using pjsip (no matter the codec).

Any help???

Regards. 


-------- Message d'origine--------
De: pjsip-bounces at lists.pjsip.org de la part de Werner Dittmann
Date: ven. 01/07/2011 16:50
?: pjsip list
Objet : Re: [pjsip] RE?:  high RTP packet loss rate when using GSM over GPRS
 
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?
> 
> 
> 
> 
> 
> 
> 
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> 
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