Codec change during call only changes receiving stream

[simon-eis]

Hi,

i have following test setup:
Microsip <-> Asterisk Server <-> Baresip
At the microsip only the codec G711A is activated, at baresip the codecs G711A, G711U, G722, G729, opus are activated.
If i establish a call from Microsip to Baresip the call is initiated correctly, but directly after the call is established i see the log output:
Audio decoder changed payload 8 -> 9
This is because baresip expects the incomming payload to be of type G711A, but it is G722. Therefore the stream_pt_handler is called and the audio player is terminated, the paramters are updated and the audio player is started again with the correct paramters. This playback seems to work, but i do not get any outgoing rtp stream from baresip (verified with wireshark). Although according to the logs my source module is active. Therefore i have some questions:

1. In the stream_pt_handler only the audio decoder is updated, does this mean, the encoder stays at the previous codec?
2. Does this change of payload influence the source pipeline?
3. Would it make sense to also update the audio encoder in the stream_pt_handler

I have already tried the fix introduced from this discussion, which did not work.

Thanks in advance.
Simon