[ENet-discuss] not enough throughput when packet loss is added

[Chris Barrett]

I have a streaming uncompressed audio application.  It is a round trip path
from node A to node B and then back to node A.  A is setup as a client.  B
is setup as a host.  The links can have up to three 48k/24bit PCM audio
channels - thus 1.1 Mbit each, plus one setup/control channel with
negligible bandwidth requirements.  This application cannot have errors, so
enet packets are all setup as reliable.

I have tried these cases:

(1) One audio channel, no packet drops, no network delay - works great.
(2) One audio channel, 0.85% packet loss in each direction, 120ms network
delay in each direction - works great with ~900ms buffer at both A and B
(3) Two audio channels, no packet drops, no network delay - works great.
(4) Two audio channels, 0.85% packet loss in each direction, 120ms network
delay in each direction - fails consistently with not enough throughput.
Thus, the buffers at either node deplete relatively quickly (~2 seconds)
and the stream is broken, falls behind.

I have set both enet hosts up for infinite bandwidth.  I believe the
network bandwidth capacity is very large, > 100 Mbit.  I setup the network
conditioner (that is simulating the packet loss and network delay) to have
60 Mbit bandwidth...so I don't think it is the problem.

At both nodes I have enet_host_service setup in it's own task, so I believe
enet is always able to perform any necessary work.

Any ideas?  I see there is some sort of throttling algorithm in enet, but
it appears to be used to limit unreliable communication (?).  Any ideas?

Thanks,
Chris
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