Softmix bridge delivering silent packets from WhatsApp (Meta) to WebRTC client

Asterisk Asterisk WebRTC
1

Hey everyone, hoping someone can shed some light on a weird situation.

I’m running Asterisk 22.8.2 with a WebRTC browser client calling out to a WhatsApp number through Meta’s SIP gateway (wa.meta.vc). Both legs negotiate opus only. This setup was working fine for a couple of days, then out of nowhere the simple bridge stopped delivering RTP packets from Meta to the WebRTC client entirely. No changes were made on my end.

I switched to softmix as a workaround and packets are now being delivered, but the audio from WhatsApp to the browser is completely silent. Occasionally there’s a pop or tap which suggests maybe 1 in a few hundred packets has actual audio content. Audio from the browser to WhatsApp works fine.

Looking at the channel formats during an active call:

WebRTC channel: NativeFormats: (opus) / WriteFormat: slin48 / ReadFormat: slin48 WriteTranscode: (slin@48000)->(opus@48000) ReadTranscode: (opus@48000)->(slin@48000)

WhatsApp channel: NativeFormats: (opus) / WriteFormat: slin / ReadFormat: slin48 WriteTranscode: (slin@8000)->(slin@48000)->(opus@48000) ReadTranscode: (opus@48000)->(slin@48000)

So softmix is mixing at 8kHz internally even though both endpoints are opus@48000. I traced this to Meta’s SDP answer which includes these fmtp parameters on the opus line:

maxplaybackrate=16000; sprop-maxcapturerate=16000; maxaveragebitrate=20000

This seems to be confusing softmix into picking 8kHz as its mixing rate. Looking at RTP packet sizes confirms it — packets sent TO WhatsApp (157.240.x.x) are tiny (~40-50 bytes, real audio) and packets sent TO the browser (190.56.x.x) are consistently 69 bytes which appears to be padded silence:

Sent RTP to 157.240.233.52 len 000046 (real audio going to WhatsApp) Sent RTP to 190.56.117.170 len 000069 (silent/padded going to browser) Sent RTP to 157.240.233.52 len 000052 Sent RTP to 190.56.117.170 len 000069 Sent RTP to 157.240.233.52 len 000039 Sent RTP to 190.56.117.170 len 000069

I’ve tried SOFTMIX_RATE=16000, BRIDGE_NATIVE_RATE=16000, and CHANNEL(audioreadformat)=slin16 in the pre-dial handler but the bridge ignores all of them — the variables are present on the channel but the mixing rate stays at 8kHz.

Is there a way to force softmix’s internal mixing rate to 16kHz or strip the maxplaybackrate/sprop-maxcapturerate from Meta’s SDP answer before it affects the bridge? Or should I try to go back to simple bridge and fix whatever issue was there (I couldn’t even determine the reason why simple bridge stopped working :/)

Thanks

2

I would start by determining why a simple bridge doesn’t work, because internally all it does is take a frame from one channel and give it to the other. If that’s not working, then it’s unlikely softmix will work - and the only reason softmix would be providing audio is because it provides a constant stream no matter what. The fact it’s doing that doesn’t mean the ingress side is working.

3

Thank you so much for your response. I have been wasting so much time on softmix when the issue was probably elsewhere.

Meta’s RTP packets ARE arriving at Asterisk correctly from 157.240.233.52. However the simple bridge is not forwarding any of them to the WebRTC client — there are zero “Sent RTP” entries to the browser IP in the logs while dozens of “Got RTP” from Meta are visible.

The logs also show this:

bridge_native_rtp.c: Bridge can not use native RTP bridge as it was forbidden while getting details

So the native RTP passthrough is being blocked for some reason, and the simple bridge falls back to… nothing, apparently. The packets just get dropped.

Any idea what would cause native RTP bridge to be forbidden in this scenario? Both legs are DTLS/SRTP with ICE — one is WSS (WebRTC client) and the other is TLS to wa.meta.vc. Could the mismatch between DTLS setups on the two legs be what’s blocking native passthrough?

4

DTLS-SRTP makes native bridging ineligible. It has to be decrypted and go through the core.

You aren’t using a patched Asterisk are you?

5

Thank you once again for your response. A team set up this Asterisk installation on an Azure server. We downloaded Asterisk from https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22-current.tar.gz but because WhatsApp requires Opus, I checked the installed files and codec_opus.so was missing. To get it working I used this repo: GitHub - traud/asterisk-opus: Asterisk 13 transcoding module: Opus . Could this be the reason for the whole mess?

6

Thank you once again for your response. A team set up this Asterisk installation (asterisk-22-curren) on an Azure server. However, because WhatsApp requires Opus, I checked the installed files and codec_opus.so was missing. To get it working I used this repo: traud/asterisk-opus. Could this be the reason for the whole mess?

7

Yes. The “enable_native_plc” patch has been known to cause audio to be dropped.

8

Thank you so much for that, really appreciate it. I went back and checked and did find the enable_native_plc patch. I’ve removed it and I believe that issue is now resolved. Not 100% sure though because I just tried another call and I’m still not getting audio from WhatsApp to my WebRTC client.

Here’s what I observed in this latest call: while the phone is ringing, Asterisk correctly sends RTP to my WebRTC frontend (the ringback beeps). Once I answer on WhatsApp, audio from my browser goes through Asterisk to the phone fine; the person on WhatsApp can hear me. But Asterisk never forwards WhatsApp’s audio back to my browser. I can see packets arriving from 157.240.233.52 in the logs, but there are no corresponding “Sent RTP” lines back to my browser IP after the call is answered.

I copied and pasted the logs below. I trimmed some parts and added comments in between sections to make it easier to follow. Sorry if it’s still messy, just trying to give as much context as possible.

I START THE CALL ON WEBRTC CLIENT...





sudo tail -f /var/log/asterisk/full | grep -i "native\|bridge\|rtp\|forbidden\|opus"

[2026-02-22 00:59:24] DEBUG[265062] res_rtp_asterisk.c: Resolved stunaddr 'stun.l.google.com' to '74.125.250.129'. Lowest TTL = 300.

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  webrtc-client

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c: Transport transport-wss bound to 0.0.0.0: Using it for RTP media.

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x72058835b110'

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) RTP allocated port 10006

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE creating session 0.0.0.0:10006 (10006)

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE create

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE add system candidates

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE add candidate: 10.0.0.15:10006, 2130706431

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE request STUN TCP RTP candidate

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE add candidate: 123.123.HIDDEN.123.123:10006, 1694498815

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: RTP instance '0x72058835b110' is setup and ready to go

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE change number of components 2 -> 1

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE resetting

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c:  (0x72058835b110) ICE nevermind, not ready for a reset

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: () RTCP setup on RTP instance

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS RTP setup

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS RTCP setup

[2026-02-22 01:00:29] DEBUG[265052] res_srtp.c: local_key64 VSULOK+bhsS9TSFyCop8JDGMtvY8zZ9wY6VhkuDM len 40

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  webrtc-client

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  webrtc-client

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 9 based on m type on 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 0 based on m type on 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 8 based on m type on 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 13 based on m type on 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session/pjsip_session_caps.c: 'webrtc-client' Caps for incoming audio call with pref 'local' - remote: (opus|g722|ulaw|alaw) local: (opus) joint: (opus)

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 0 (0x72058801d528) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 8 (0x7205880ec078) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 9 (0x72058804b948) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 13 (0x61be96d58818) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 110 (0x7205880ef178) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 111 (0x72058831c078) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Crossover copying tx to rx payload mapping 126 (0x7205880dcda8) from 0x720567bfe120 to 0x720567bfe120

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 0 (0x72058801d528) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 8 (0x7205880ec078) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 9 (0x72058804b948) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 13 (0x61be96d58818) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 110 (0x7205882fb448) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 111 (0x72058831c078) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying rx payload mapping 126 (0x7205880dcda8) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 0 (0x72058801d528) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 8 (0x7205880ec078) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 9 (0x72058804b948) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 13 (0x61be96d58818) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 110 (0x7205880ef178) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 111 (0x72058831c078) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 126 (0x7205880dcda8) from 0x720567bfe120 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  webrtc-client: Media stream 0:audio-0:audio:sendrecv (opus) handled by audio

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  webrtc-client: Done with stream 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  webrtc-client: Processing stream 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  webrtc-client Stream: 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  webrtc-client Type: audio 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) RTCP ignoring duplicate property

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS RTP setup

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  RC: 1

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  webrtc-client: Stream 0:audio-0:audio:sendrecv (opus) added with mid 0

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  webrtc-client: Done with 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:29] DEBUG[265052] chan_pjsip.c:  Topology:  <0:audio-0:audio:sendrecv (opus)> Formats: (opus)

[2026-02-22 01:00:29] DEBUG[265052] channel_internal_api.c:  PJSIP/webrtc-client-00000014: MultistreamFormats: (opus)

[2026-02-22 01:00:29] DEBUG[265052] channel_internal_api.c:  Set native formats but not topology

[2026-02-22 01:00:29] DEBUG[265052] channel_internal_api.c:  PJSIP/webrtc-client-00000014:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/webrtc-client-00000014 Stream: 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) RTCP ignoring duplicate property

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS RTP setup

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) RTCP setting address on RTP instance

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c: (0x72058835b110) ICE process attributes

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE add remote candidate

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE add remote candidate

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE add remote candidate

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE set role to CONTROLLED

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE start

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE resetting

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c:  (0x72058835b110) ICE nevermind, not ready for a reset

[2026-02-22 01:00:29] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE successfully created checklist

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/webrtc-client-00000014 ANSWER

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/webrtc-client-00000014

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 9 based on m type on 0x720567bfe020

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 0 based on m type on 0x720567bfe020

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 8 based on m type on 0x720567bfe020

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Setting tx payload type 13 based on m type on 0x720567bfe020

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 0 (0x72058831af28) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 8 (0x7205880dc4a8) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 9 (0x72058801d8d8) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 13 (0x61be96d58818) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 110 (0x7205880ffb78) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 111 (0x720588286a68) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 126 (0x7205880ea7b8) from 0x720567bfe020 to 0x72058835b2e8

[2026-02-22 01:00:29] DEBUG[265052] channel_internal_api.c:  PJSIP/webrtc-client-00000014: MultistreamFormats: (opus)

[2026-02-22 01:00:29] DEBUG[265052] channel_internal_api.c:  Set native formats but not topology

[2026-02-22 01:00:29] DEBUG[265052] channel.c: Channel PJSIP/webrtc-client-00000014 setting read format path: opus -> opu

[2026-02-22 01:00:29] DEBUG[265052] channel.c: Channel PJSIP/webrtc-client-00000014 setting write format path: opus -> opus

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_sdp_rtp.c:  Handled

[2026-02-22 01:00:29] DEBUG[265052] channel_internal_api.c:  PJSIP/webrtc-client-00000014:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:29] DEBUG[265052] stream.c:  Topology: 0x7205882e8718:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:29] DEBUG[265052] res_pjsip_session.c:  Topology: Pending: (null topology)  Active:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:30] DEBUG[265052] res_pjsip_session.c:  Topology: Pending: (null topology)  Active:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) ICE valid pair, start media

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) RTCP setting address on RTP instance

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) ICE starting media - perform DTLS - (0x720588139480)

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x720588139480) DTLS perform handshake - ssl = 0x720588055440, setup = 0

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - scheduled timeout timer for '999' RTP

[2026-02-22 01:00:31] DEBUG[265066] res_rtp_asterisk.c: (0x72058835b110) ICE complete, start media

[2026-02-22 01:00:31] DEBUG[265066] res_rtp_asterisk.c: (0x72058835b110) RTCP setting address on RTP instance

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] chan_pjsip.c:  whatsapp/sip:+PHONENUMBER@wa.meta.vc Topology:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[265052] res_pjsip_session.c:  whatsapp (null) Topology:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[265052] res_pjsip_session/pjsip_session_caps.c: 'whatsapp' Caps for outgoing audio call with pref 'local_first' - remote: (opus) local: (opus) joint: (opus)

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] chan_pjsip.c:  Topology:  <0:audio-0:audio:sendrecv (opus)> Formats: (opu)

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] channel_internal_api.c:  PJSIP/whatsapp-00000015: MultistreamFormats: (opus)

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] channel_internal_api.c:  Set native formats but not topology

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] channel_internal_api.c:  PJSIP/whatsapp-00000015:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] stream.c:  Topology: 0x720570015918:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[266769][C-0000000b] chan_pjsip.c:  PJSIP/whatsapp-00000015 Topology:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[265052] chan_pjsip.c:  PJSIP/whatsapp-00000015 Topology:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:31] DEBUG[265052] res_pjsip_session.c:  PJSIP/whatsapp-00000015: Processing stream 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:31] DEBUG[265052] res_pjsip_session.c:  PJSIP/whatsapp-00000015 Stream: 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:31] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/whatsapp-00000015 Type: audio 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:31] DEBUG[265052] res_pjsip_sdp_rtp.c: Transport transport-tls bound to 0.0.0.0: Using it for RTP media.

[2026-02-22 01:00:31] DEBUG[265052] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x72058827ac50'

[2026-02-22 01:00:31] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) RTP allocated port 10016

[2026-02-22 01:00:31] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE creating session 0.0.0.0:10016 (10016)

[2026-02-22 01:00:31] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE create

[2026-02-22 01:00:31] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE add system candidates

[2026-02-22 01:00:31] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE add candidate: 10.0.0.15:10016, 2130706431

[2026-02-22 01:00:31] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE request STUN TCP RTP candidate

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE add candidate: 123.123.HIDDEN.123.123:10016, 1694498815

[2026-02-22 01:00:32] DEBUG[265052] rtp_engine.c: RTP instance '0x72058827ac50' is setup and ready to go

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE change number of components 2 -> 1

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE resetting

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c:  (0x72058827ac50) ICE nevermind, not ready for a reset

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c: () RTCP setup on RTP instance

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) DTLS RTP setup

[2026-02-22 01:00:32] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) DTLS RTCP setup

[2026-02-22 01:00:32] DEBUG[265052] res_srtp.c: local_key64 CBrvNpief0qaHkYQk5HVha2r19Bgpy0hpnVKdOs8 len 40

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_sdp_rtp.c:  RC: 1

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  PJSIP/whatsapp-00000015: Stream 0:audio-0:audio:sendrecv (opus) added with mid audio-0

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  PJSIP/whatsapp-00000015: Done with 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  Topology: Pending:  <0:audio-0:audio:sendrecv (opus)>  Active: (null topology)

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/whatsapp-00000015 setting read format path: opus -> opus

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting write format path: opus -> opus

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting read format path:opus -> opus

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/whatsapp-00000015 setting write format path: opu -> opus

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS - __rtp_recvfrom rtp=0x720588139480 - Got SSL packet '22'

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - scheduled timeout timer for '999' RTP

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  Topology: Pending:  <0:audio-0:audio:sendrecv (opus)>  Active: (null topology)

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  Topology: Pending:  <0:audio-0:audio:sendrecv (opus)>  Active: (null topology)

[2026-02-22 01:00:32] DEBUG[265057] res_pjsip_session.c:  Topology: Pending:  <0:audio-0:audio:sendrecv (opus)>  Active: (null topology)

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  Topology: Pending:  <0:audio-0:audio:sendrecv (opus)>  Active: (null topology)

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting write format path: slin -> opus

[2026-02-22 01:00:32] DEBUG[265052] res_pjsip_session.c:  Topology: Pending:  <0:audio-0:audio:sendrecv (opus)>  Active: (null topology)

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (1771722029.30) RTP ooh, format changed from none to opus

[2026-02-22 01:00:32] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (1771722029.30) RTCP starting transmission in 5000 ms

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024002, ts 000960, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024003, ts 001920, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024004, ts 002880, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024005, ts 003840, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024006, ts 004800, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024007, ts 005760, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024008, ts 006720, len -000012)

[2026-02-22 01:00:32] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024009, ts 007680, len -000012

[2026-02-22 01:00:33] DEBUG[265084] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - handle timeout - rtcp=0 result: 1

[2026-02-22 01:00:33] DEBUG[265084] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - handle timeout - rtcp=0 timeout=1999

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024040, ts 037440, len -000012)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024041, ts 038400, len -000012)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024042, ts 039360, len -000012)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024043, ts 040320, len -000012)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (type 111, seq 024044, ts 041280, len -000012)

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS - __rtp_recvfrom rtp=0x720588139480 - Got SSL packet '20'

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS setup SRTP rtp=0x720588139480'

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - add local ssrc - rtcp=0, set_remote_policy=1'

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_srtp.c: Adding new policy for SSRC 154795736

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS - __rtp_recvfrom rtp=0x720588139480 - established'

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029269, ts 3117514633, len 000661)

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: Seed ts: 3117514633 current time: 1771722033.403003

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029270, ts 3117515593, len 000661)

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: Packet 2 < 15.  Ignoring

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029271, ts 3117516553, len 000661)

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: Packet 3 < 15.  Ignoring

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029272, ts 3117517513, len 000661)

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: Packet 4 < 15.  Ignoring

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024045, ts 042240, len 000652)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029273, ts 3117518473, len 000661)

[2026-02-22 01:00:33] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: Packet 5 < 15.  Ignoring









THE FOLLOWING "Sent RTP packet to      [BROWSER-IP]:57309" ARE "BEEP BEEP" NOISES SENT FROM ASTERISK TO MY WEBRTC CLIENT. HAVENT ANSWERED YET.









[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024046, ts 043200, len 000677)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024047, ts 044160, len 000636)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024048, ts 045120, len 000633)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024049, ts 046080, len 000707)

[2026-02-22 01:00:33] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024050, ts 047040, len 000616)



...



[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: Packet 14 < 15.  Ignoring

[2026-02-22 01:00:34] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029283, ts 3117528073, len 000661)

[2026-02-22 01:00:34] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029285, ts 3117529993, len 000661)

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:    17 Arrival sec:   0.000  Arrival ts:         18  RX ts: 3117529993 Transit samp: 1177437321 Last transit samp: 1177438272 d:  951 Curr jitter:      56(  0.001) Prev Jitter:      59(  0.001) New Jitter:     115(  0.002)

[2026-02-22 01:00:34] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029286, ts 3117530953, len 000661)









SOME DTLS SET UP







[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS - __rtp_recvfrom rtp=0x720588139480 - Got SSL packet '20'

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS setup SRTP rtp=0x720588139480'

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - add local ssrc - rtcp=0, set_remote_policy=1'

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_srtp.c: Adding new policy for SSRC 154795736

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: (0x72058835b110) DTLS - __rtp_recvfrom rtp=0x720588139480 - established'

[2026-02-22 01:00:34] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029318, ts 3117561673, len 000661)

[2026-02-22 01:00:34] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:    50 Arrival sec:   0.178  Arrival ts:       8538  RX ts: 3117561673 Transit samp: 1177414161 Last transit









BACK TO BEEP BEEP NOISE ON THE WEBRTC FRONT END







samp: 1177414992 d:  831 Curr jitter:      -7(89478.485) Prev Jitter:     936(  0.019) New Jitter:     929(  0.019)

[2026-02-22 01:00:34] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024090, ts 085440, len 000663)

[2026-02-22 01:00:34] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029319, ts 3117562633, len 000661)



...



[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   175 Arrival sec:   2.687  Arrival ts:     128979  RX ts: 3117681673 Transit samp: 1177414602 Last transit samp: 1177415561 d:  959 Curr jitter:     -22(89478.485) Prev Jitter:    1313(  0.027) New Jitter:    1290(  0.027)

[2026-02-22 01:00:36] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      [BROWSER-IP]:57309 (via ICE) (type 111, seq 024216, ts 206400, len 000013)
9

And here is the rest. Thank you so much for your help :folded_hands:

HERE I PICK UP THE CALL ON MY WHATSAPP



[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/whatsapp-00000015 Stream: 0:audio-0:audio:sendrecv (opus)

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) RTCP ignoring duplicate property

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) DTLS RTP setup

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) RTCP setting address on RTP instance

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c: (0x72058827ac50) ICE process attributes

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE add remote candidate

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE add remote candidate

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE set role to CONTROLLING

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE start

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE resetting

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c:  (0x72058827ac50) ICE nevermind, not ready for a reset

[2026-02-22 01:00:36] DEBUG[265052] res_rtp_asterisk.c: (0x72058827ac50) ICE successfully created checklist

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/whatsapp-00000015 ANSWER

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c:  PJSIP/whatsapp-00000015

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:36] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 102 (0x7205881de078) from 0x720567bfde10 to 0x72058827ae28

[2026-02-22 01:00:36] DEBUG[265052] rtp_engine.c: Copying tx payload mapping 111 (0x7205881f8cd8) from 0x720567bfde10 to 0x72058827ae28

[2026-02-22 01:00:36] DEBUG[265052] channel_internal_api.c:  PJSIP/whatsapp-00000015: MultistreamFormats: (opus)

[2026-02-22 01:00:36] DEBUG[265052] channel_internal_api.c:  Set native formats but not topology

[2026-02-22 01:00:36] DEBUG[265052] channel.c: Channel PJSIP/whatsapp-00000015 setting read format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[265052] channel.c: Channel PJSIP/whatsapp-00000015 setting write format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c:

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_sdp_rtp.c:  Handled

[2026-02-22 01:00:36] DEBUG[265052] channel_internal_api.c:  PJSIP/whatsapp-00000015:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[265052] stream.c:  Topology: 0x720570026df8:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[265052] res_pjsip_session.c:  Topology: Pending: (null topology)  Active:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting write format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/whatsapp-00000015 setting read format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting write format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting read format path:opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/whatsapp-00000015 setting write format path: opu -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis.c: Creating topic. name: bridge:all/bridge:287736cc-034c-4600-8fdf-276e0e974e5e, detail:

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis.c: Topic 'bridge:all/bridge:287736cc-034c-4600-8fdf-276e0e974e5e': 0x720570077350 created

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): base_init

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge_native_rtp.c: Bridge '287736cc-034c-4600-8fdf-276e0e974e5e' can not use native RTP bridge as two channels are required

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge technology softmix has less preference than simple_bridg (10 <= 50). Skipping.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Chose bridge technology simple_bridge

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: calling simple_bridge technology constructor

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: calling simple_bridge technology start

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): base_init complete

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): registering

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x7205700585f8  Old: <none>  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x7205700585f8  Old: <none>  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: 0x720570058f20(PJSIP/whatsapp-00000015) is joining

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: pushing 0x720570058f20(PJSIP/whatsapp-00000015)

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c0216c8  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c0216c8  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge_native_rtp.c: Bridge '287736cc-034c-4600-8fdf-276e0e974e5e' can not use native RTP bridge as two channels are required

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge.c: Bridge technology softmix does not have any capabilities we want.

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge.c: Chose bridge technology simple_bridge

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e is already using the new technology.

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: 0x720570058f20(PJSIP/whatsapp-00000015) is joining simple_bridge technology

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c020448  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c020448  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: 0x72057007dc50(PJSIP/webrtc-client-00000014) is joining

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: pushing 0x72057007dc50(PJSIP/webrtc-client-00000014)

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x720570074c88  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x720570074c88  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge_native_rtp.c: Bridge '287736cc-034c-4600-8fdf-276e0e974e5e'.  Checking compatability for channels 'PJSIP/whatsapp-00000015' and 'PJSIP/webrtc-client-00000014'

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge_native_rtp.c: Bridge '287736cc-034c-4600-8fdf-276e0e974e5e' can not use native RTP bridge as it was forbidden while getting details

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge technology native_rtp is not compatible with properties of existing bridge.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge technology holding_bridge does not have any capabilities we want.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge technology softmix does not have any capabilities we want.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Chose bridge technology simple_bridge

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e is already using the new technology.

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: 0x72057007dc50(PJSIP/webrtc-client-00000014) is joining simple_bridge technology

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting read format path:opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/whatsapp-00000015 setting write format path: opu -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/whatsapp-00000015 setting read format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: Channel PJSIP/webrtc-client-00000014 setting write format path: opus -> opus

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] bridge_simple.c: PJSIP/webrtc-client-00000014: Topologies already match. Current:  <0:audio-0:audio:sendrecv (opus)>  Requested:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stream.c:  Topology: 0x72057006bc98:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] channel.c: PJSIP/whatsapp-00000015: Topologies already match. Current:  <0:audio-0:audio:sendrecv (opus)>  Requested:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stream.c:  Topology: 0x72057006bc98:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x720570073a78  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x720570073a78  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:36] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029444, ts 3117682633, len 000661)

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   176 Arrival sec:   2.729  Arrival ts:     130977  RX ts: 3117682633 Transit samp: 1177415640 Last transit samp: 1177414602 d: 1038 Curr jitter:     -16(89478.485) Prev Jitter:    1290(  0.027) New Jitter:    1275(  0.027)

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (1771722031.31) RTP ooh, format changed from none to opus

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (1771722031.31) RTCP starting transmission in 5000 ms

[2026-02-22 01:00:36] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016969, ts 3117682608, len -000012)

[2026-02-22 01:00:36] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029445, ts 3117683593, len 000662)

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   177 Arrival sec:   2.735  Arrival ts:     131270  RX ts: 3117683593 Transit samp: 1177414973 Last transit samp: 1177415640 d:  667 Curr jitter:     -38(89478.485) Prev Jitter:    1275(  0.027) New Jitter:    1237(  0.026)

[2026-02-22 01:00:36] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029446, ts 3117684553, len 000660)

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   178 Arrival sec:   2.735  Arrival ts:     131272  RX ts: 3117684553 Transit samp: 1177414015 Last transit samp: 1177414973 d:  958 Curr jitter:     -17(89478.485) Prev Jitter:    1237(  0.026) New Jitter:    1219(  0.025)

[2026-02-22 01:00:36] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016970, ts 3117683568, len -000012)

[2026-02-22 01:00:36] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016971, ts 3117684528, len -000012)

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (0x72058827ac50) ICE valid pair, start media

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (0x72058827ac50) RTCP setting address on RTP instance

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (0x72058827ac50) ICE starting media - perform DTLS - (0x720588145f90)

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (0x720588145f90) DTLS perform handshake - ssl = 0x720588149810, setup = 0

[2026-02-22 01:00:36] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (0x72058827ac50) DTLS srtp - scheduled timeout timer for '999' RTP

[2026-02-22 01:00:36] DEBUG[265066] res_rtp_asterisk.c: (0x72058827ac50) ICE complete, start media

[2026-02-22 01:00:36] DEBUG[265066] res_rtp_asterisk.c: (0x72058827ac50) RTCP setting address on RTP instance











BROWSER STARTS SENDING AUDIO TO PHONE, BUT NOT THE OTHER WAY AROUND





[2026-02-22 01:00:36] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029447, ts 3117685513, len 000662)

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   179 Arrival sec:   2.859  Arrival ts:     137219  RX ts: 3117685513 Transit samp: 1177419002 Last transit samp: 1177414015 d: 4987 Curr jitter:     235(  0.005) Prev Jitter:    1219(  0.025) New Jitter:    1455(  0.030)

[2026-02-22 01:00:36] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029448, ts 3117686473, len 000660)

[2026-02-22 01:00:36] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   180 Arrival sec:   2.859  Arrival ts:     137238  RX ts: 3117686473 Transit samp: 1177418061 Last transit samp: 1177419002 d:  941 Curr jitter:     -32(89478.485) Prev Jitter:    1455(  0.030) New Jitter:    1423(  0.030)

[2026-02-22 01:00:36] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016972, ts 3117685488, len -000012)

[2026-02-22 01:00:36] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016973, ts 3117686448, len -000012)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029449, ts 3117687433, len 000661)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   181 Arrival sec:   2.867  Arrival ts:     137594  RX ts: 3117687433 Transit samp: 1177417457 Last transit samp: 1177418061 d:  604 Curr jitter:     -51(89478.485) Prev Jitter:    1423(  0.030) New Jitter:    1372(  0.029)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029450, ts 3117688393, len 000661)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   182 Arrival sec:   2.867  Arrival ts:     137624  RX ts: 3117688393 Transit samp: 1177416527 Last transit samp: 1177417457 d:  930 Curr jitter:     -28(89478.485) Prev Jitter:    1372(  0.029) New Jitter:    1344(  0.028)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029451, ts 3117689353, len 000661)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   183 Arrival sec:   2.867  Arrival ts:     137627  RX ts: 3117689353 Transit samp: 1177415570 Last transit samp: 1177416527 d:  957 Curr jitter:     -24(89478.485) Prev Jitter:    1344(  0.028) New Jitter:    1320(  0.027)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029452, ts 3117690313, len 000661)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   184 Arrival sec:   2.867  Arrival ts:     137628  RX ts: 3117690313 Transit samp: 1177414611 Last transit samp: 1177415570 d:  959 Curr jitter:     -23(89478.485) Prev Jitter:    1320(  0.027) New Jitter:    1297(  0.027)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016974, ts 3117687408, len -000012)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016975, ts 3117688368, len -000012)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016976, ts 3117689328, len -000012)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (type 111, seq 016977, ts 3117690288, len -000012)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029453, ts 3117691273, len 000661)



...



[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   442 Arrival sec:   8.109  Arrival ts:     389236  RX ts: 3117937993 Transit samp: 1177418539 Last transit samp: 1177419360 d:  821 Curr jitter:     -22(89478.485) Prev Jitter:    1170(  0.024) New Jitter:    1148(  0.024)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029711, ts 3117938953, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   443 Arrival sec:   8.109  Arrival ts:     389246  RX ts: 3117938953 Transit samp: 1177417589 Last transit samp: 1177418539 d:  950 Curr jitter:     -12(89478.485) Prev Jitter:    1148(  0.024) New Jitter:    1136(  0.024)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029712, ts 3117939913, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   444 Arrival sec:   8.110  Arrival ts:     389257  RX ts: 3117939913 Transit samp: 1177416640 Last transit samp: 1177417589 d:  949 Curr jitter:     -12(89478.485) Prev Jitter:    1136(  0.024) New Jitter:    1124(  0.023)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017234, ts 3117937008, len 000661)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017235, ts 3117937968, len 000661)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017236, ts 3117938928, len 000661)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029713, ts 3117940873, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   445 Arrival sec:   8.110  Arrival ts:     389276  RX ts: 3117940873 Transit samp: 1177415699 Last transit samp: 1177416640 d:  941 Curr jitter:     -11(89478.485) Prev Jitter:    1124(  0.023) New Jitter:    1113(  0.023)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017237, ts 3117939888, len 000661)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017238, ts 3117940848, len 000661)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029714, ts 3117941833, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   446 Arrival sec:   8.110  Arrival ts:     389290  RX ts: 3117941833 Transit samp: 1177414753 Last transit samp: 1177415699 d:  946 Curr jitter:     -10(89478.485) Prev Jitter:    1113(  0.023) New Jitter:    1102(  0.023)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017239, ts 3117941808, len 000661)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029715, ts 3117942793, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   447 Arrival sec:   8.151  Arrival ts:     391262  RX ts: 3117942793 Transit samp: 1177415765 Last transit samp: 1177414753 d: 1012 Curr jitter:      -6(89478.485) Prev Jitter:    1102(  0.023) New Jitter:    1097(  0.023)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029716, ts 3117943753, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   448 Arrival sec:   8.152  Arrival ts:     391274  RX ts: 3117943753 Transit samp: 1177414817 Last transit samp: 1177415765 d:  948 Curr jitter:      -9(89478.485) Prev Jitter:    1097(  0.023) New Jitter:    1087(  0.023)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017240, ts 3117942768, len 000661)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017241, ts 3117943728, len 000661)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029717, ts 3117944713, len 000661)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   449 Arrival sec:   8.157  Arrival ts:     391513  RX ts: 3117944713 Transit samp: 1177414096 Last transit samp: 1177414817 d:  721 Curr jitter:     -23(89478.485) Prev Jitter:    1087(  0.023) New Jitter:    1065(  0.022)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017242, ts 3117944688, len 000661)

[2026-02-22 01:00:42] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got  RTP packet from    [BROWSER-IP]:57309 (type 111, seq 029718, ts 3117945673, len 000661)



...



[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt:   456 Arrival sec:   8.771  Arrival ts:     420984  RX ts: 3117951433 Transit samp: 1177436847 Last transit samp: 1177413918 d: 22929 Curr jitter:    1362(  0.028) Prev Jitter:    1129(  0.024) New Jitter:    2492(  0.052)

[2026-02-22 01:00:42] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to      157.240.233.52:3484 (via ICE) (type 111, seq 017249, ts 3117951408, len 000661)





HERE I HUNG UP THE CALL





[2026-02-22 01:00:42] DEBUG[265052] res_pjsip_session.c:  Topology: Pending: (null topology)  Active:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:42] DEBUG[265052] res_pjsip_session.c:  Topology: Pending: (null topology)  Active:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge_channel.c: Setting 0x72057007dc50(PJSIP/webrtc-client-00000014) state from:0 to:1

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: pulling 0x72057007dc50(PJSIP/webrtc-client-00000014)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: 0x72057007dc50(PJSIP/webrtc-client-00000014) is leaving simple_bridge technology

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): dissolving with cause 16(Normal Clearing)

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): kicking channel PJSIP/whatsapp-00000015

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge_channel.c: Setting 0x720570058f20(PJSIP/whatsapp-00000015) state from:0 to:2

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: queueing action type:13 sub:1001

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): DEFERRED_DISSOLVING queued.  current refcound: 5

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x7205700750d8  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] stasis_bridges.c: Update: 0x7205700750d8  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:42] DEBUG[266769][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e is dissolved, not performing smart bridge operation.

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: pulling 0x720570058f20(PJSIP/whatsapp-00000015)

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge_channel.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: 0x720570058f20(PJSIP/whatsapp-00000015) is leaving simple_bridge technology

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c087778  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c087778  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: 287736cc-034c-4600-8fdf-276e0e974e5e

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e is dissolved, not performing smart bridge operation.

[2026-02-22 01:00:42] DEBUG[265039][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): unlinking bridge.  Refcount: 3

[2026-02-22 01:00:42] DEBUG[265039][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): unlinked bridge.  Refcount: 2

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): actually destroying basic bridge, nobody wants it anymore

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis_bridges.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): destroying topics

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c02bd88  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: <none>

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis_bridges.c: Update: 0x72057c02bd88  Old: 287736cc-034c-4600-8fdf-276e0e974e5e  New: <none>

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis.c: Destroying topic. name: bridge:all/bridge:287736cc-034c-4600-8fdf-276e0e974e5e, detail:

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] stasis.c: Topic 'bridge:all/bridge:287736cc-034c-4600-8fdf-276e0e974e5e': 0x720570077350 destroyed

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: calling basic bridge destructor

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): destroying bridge (noop)

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: calling simple_bridge technology stop

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e: calling simple_bridge technology destructor

[2026-02-22 01:00:42] DEBUG[266770][C-0000000b] bridge.c: Bridge 287736cc-034c-4600-8fdf-276e0e974e5e(<unknown>)(0x72057006e190): destroyed

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (1771722029.30) RTP Stop

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS stop

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE RTP transport deallocating

[2026-02-22 01:00:42] DEBUG[265052] res_rtp_asterisk.c: (0x72058835b110) ICE stopped

[2026-02-22 01:00:42] DEBUG[265052] rtp_engine.c: Destroyed RTP instance '0x72058835b110'

[2026-02-22 01:00:42] DEBUG[265052] stream.c:  Topology: 0x7205880aefd8:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:42] DEBUG[265052] stream.c:  Topology: 0x720588054eb8:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (1771722031.31) RTP Stop

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) DTLS stop

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) DTLS srtp - stopped timeout timer'

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) ICE RTP transport deallocating

[2026-02-22 01:00:42] DEBUG[265053] res_rtp_asterisk.c: (0x72058827ac50) ICE stopped

[2026-02-22 01:00:42] DEBUG[265053] rtp_engine.c: Destroyed RTP instance '0x72058827ac50'

[2026-02-22 01:00:42] DEBUG[265053] stream.c:  Topology: 0x72058831beb8:  <0:audio-0:audio:sendrecv (opus)>

[2026-02-22 01:00:42] DEBUG[265053] stream.c:  Topology: 0x7205881f8d28:  <0:audio-0:audio:sendrecv (opus)>
10

…where?

I see no packets being received at the RTP level from that address in the logs you’ve provided.

11

Oop, sorry. I think I trimmed the logs too much. Here is the part where you can see the incoming ones from Meta. I’ve noticed that they are really small packages; always between len 000020 and len 000040 in every call I make. This despite me having both my computer and phone next to me and both picking up the same audio.

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000034, ts 160320, len 000040)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000035, ts 161280, len 000033)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 190.56.117.170:57309 (type 111, seq 029456, ts 3117694153, len 000661)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt: 188 Arrival sec: 2.994 Arrival ts: 143699 RX ts: 3117694153 Transit samp: 1177416842 Last transit samp: 1177414264 d: 2578 Curr jitter: 89( 0.002) Prev Jitter: 1152( 0.024) New Jitter: 1241( 0.026)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 190.56.117.170:57309 (type 111, seq 029457, ts 3117695113, len 000671)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt: 189 Arrival sec: 2.994 Arrival ts: 143713 RX ts: 3117695113 Transit samp: 1177415896 Last transit samp: 1177416842 d: 946 Curr jitter: -18(89478.485) Prev Jitter: 1241( 0.026) New Jitter: 1222( 0.025)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to 157.240.233.52:3484 (via ICE) (type 111, seq 016981, ts 3117694128, len 000661)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to 157.240.233.52:3484 (via ICE) (type 111, seq 016982, ts 3117695088, len 000671)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 190.56.117.170:57309 (type 111, seq 029458, ts 3117696073, len 000651)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt: 190 Arrival sec: 2.994 Arrival ts: 143729 RX ts: 3117696073 Transit samp: 1177414952 Last transit samp: 1177415896 d: 944 Curr jitter: -17(89478.485) Prev Jitter: 1222( 0.025) New Jitter: 1205( 0.025)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to 157.240.233.52:3484 (via ICE) (type 111, seq 016983, ts 3117696048, len 000651)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 190.56.117.170:57309 (type 111, seq 029459, ts 3117697033, len 000665)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt: 191 Arrival sec: 3.002 Arrival ts: 144073 RX ts: 3117697033 Transit samp: 1177414336 Last transit samp: 1177414952 d: 616 Curr jitter: -37(89478.485) Prev Jitter: 1205( 0.025) New Jitter: 1168( 0.024)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Sent RTP packet to 157.240.233.52:3484 (via ICE) (type 111, seq 016984, ts 3117697008, len 000665)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000036, ts 162240, len 000045)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000037, ts 163200, len 000034)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000038, ts 164160, len 000033)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000039, ts 165120, len 000032)

[2026-02-22 01:00:37] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: (1771722031.31) RTCP got report of 52 bytes from 157.240.233.52:3484

[2026-02-22 01:00:37] DEBUG[266770][C-0000000b] res_rtp_asterisk.c: 1771722031.31: rtt: 0.000000000 j: 0.000000000 sjh: 0.000000000 lost: 0.000000000 mes: 88.1

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000040, ts 166080, len 000029)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000041, ts 167040, len 000037)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000042, ts 168000, len 000043)

[2026-02-22 01:00:37] VERBOSE[266770][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 157.240.233.52:3484 (type 111, seq 000043, ts 168960, len 000033)

[2026-02-22 01:00:37] VERBOSE[266769][C-0000000b] res_rtp_asterisk.c: Got RTP packet from 190.56.117.170:57309 (type 111, seq 029460, ts 3117697993, len 000677)

[2026-02-22 01:00:37] DEBUG[266769][C-0000000b] res_rtp_asterisk.c: 1771722029.30: pkt: 192 Arrival sec: 3.135 Arrival ts: 150484 RX ts: 3117697993 Transit samp: 1177419787 Last transit samp: 1177414336 d: 5451 Curr jitter: 268( 0.006) Prev Jitter: 1168( 0.024) New Jitter: 1436( 0.030)

12

Yes, they are quite small. I don’t know why, ‘nor do I know why it’s not working. I don’t have anything else to add.

13

Hi @stefanogt are you able to share a pcap for the leg between Meta and Asterisk?