I am using a Media GWY with Asterisk. Asterisk interfaces to the PSTN using TE420 card.
Phones <—>3rd party Media GWY<----SIP/RTP---->Asterisk<–E1->PSTN
I can make and receive call to from the PSTN without any problem. But when a DTMF is pressed, the call quality degrades and both ends start hearing garbled and choppy voice. Farther investigation showed that when a DTMF is pressed, the 3rd party Media GWY continue to send RTP packets to Asterisk but Asterisk pauses to send RTP packets to the media gwy.
However, when RTP packets resume the sequence numbering of the packets continues from the next sequence number rather than leaving a hole in the sequencing to represent the packets that were never sent. This results in skew between the sequence numbers and the times-of-arrival in the RTP packets. The medie gwy concludes (incorrectly) that these packets are late and, depending on the size of the jitter buffer, either drops them all (resulting in silence) or accepts only those packets whose arrival times it thinks are late by less than the size of the jitter buffer (resulting in choppy voice ).
Is it possible to configure Asterisk to continue sending RTP packets during a DTMF press ? Or to increment the sequence number by the number of silence packets ?