Problems with Opus, Grandstream HT802, directmedia, native_rtp

Asterisk Asterisk Support
5

Here’s some interesting output from the Grandstream HT802 syslog:

...
Jan 20 08:46:30 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: New Codec '0' Rx_pt '0' iLoop '1'
Jan 20 08:46:30 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS:  #### Received the RTP Packet PT 0 which is different of what was configured 123
Jan 20 08:46:30 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS:  The function p_rtp_PacketCheck returned NULL OR NSE Pkt recv
Jan 20 08:46:30 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS:  Received codec is 0 and expected is 123
...
Jan 20 08:46:34 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS:  @@@##  #$$$  Sending RTCP  based on TS
Jan 20 08:46:34 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: Sending RTCP pkt type '200' of length '52' to user space !!!
Jan 20 08:46:34 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: Sending RTCP pkt type '202' of length '20' to user space !!!
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS:  RTCP pkt received
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: RDRD Cordless: Receiving RTCP 1
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: error2
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: RTCP packet received: 200 error!!!
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: error2
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: RTCP packet received: 202 error!!!
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: p_rtp_SeqInit 291
Jan 20 08:46:35 HT802 [00: 0B:82:96:AE:C2] [1.0.2.5] LIBGSDSP: CSS: p_rtp_SeqInit 293
...

It seems to not like receiving ulaw while it is sending Opus.

Edit:

I found this post on the Grandstream forums: http://forums.grandstream.com/forums/index.php?topic=33128.0

Does Asterisk / PJSIP allow for asymmetric codecs? I am finding some old posts on mailing lists suggesting it does not, but maybe PJSIP allows for this.

Is there a way to turn this off without completely disabling native_rtp bridges?

I have:

asymmetric_rtp_codec=false

for all endpoints.