I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration:
SIP <-----> Asterisk 1 <----> Asterisk 2 <----> Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2. I've read the source code of chan_dahdi, and I think the channel has a 20 ms "buffer" (160 samples). Algorithms like mg2 and kb1 are configured to accept 128 taps (16 ms), so 20 ms is too high. Someone knows how I can reduce the delay to at least 10 ms? Should I change something in the source code? Thanks in advance, Gustavo Santos. -- Atenciosamente, Gustavo Santos.
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