too many packets lost in a single rtp stream

[nabi]

Hi all,

I am trying to set an very simple experiment for VoIP purposes. The system set up is the following. One computer (User1) is on a Ethenert network connected through a Full duplex to an Wifi (802.11b) Access Point(AP). In the wifi AP there is only one computer (User2) whose aim is to send RTP packets to User1 and listen to those that are coming back. From User1 prespective, it will receive packets from User2 and send them back to User2, as a proxy style.

--------Wifi-------------Eth
User2<------->AP<--------->User1

SendtoUser1(P1):
ReceivePacketfromUser2(P1);
ForwardPackettoUser1(P1);
ReceivePacketfromUser1(P1);

Now, I have wireshark running in both machines, User1 and User2, plus i check the sequence number of RTP packet to make sure that i know when a packet is received. Surprisingly for a packet flow of 706 packets, User2 rarely gets more than 650 packets back. Since Wireshark of User 1 shows that all 706 packets were sent back I can guess that packet were lost at the AP buffer.

Question is, is it not too many packets lost for a single RTP streaming transmission? I could expect to loose some of the packet but not a 8% or so.
The wireless router that i am using is a Dlink G604.

Thanks

[Salem]

Are you saturating the channel with messages in a tight loop, or are you modelling a realistic data flow?

[nabi]

are you modelling a realistic data flow?

I am using ortp libraries and two linux computers.

"Are you saturating the channel with messages in a tight loop"

Well, I am sending one RTPstream from user2 to user1 and another stream from user2 to user1with the particularity that the second stream is a copy of the first one. I am loosing packets, but the question is, Am i really saturating an AP by only two RTPstreams? that looks to me a poor performance.

[Salem]

I mean, if your video stream is meant to be viewed in real time over a period of 1 hour, are you sending it a few packets at a time over a 1 hour period?

Or are you attempting to send the whole lot in 5 minutes. This is the saturation test, and is NOT what RTP is designed to support with almost zero packet loss.

[nabi]

right,

I am sending packets and receiving them in real time, there is no buffering. Take it as a VoIP call where i send and receive my own data. I understand this is what you are calling a Saturation test.

[Salem]

Well it would seem to me that you're using RTP in a manner which it wasn't designed for.