salmon-leather-15461
08/06/2025, 12:21 PMBVCTelephony
agent code. However, we're still noticing that the bot picks up its own welcome message. We're not sure why this is happening. Any Suggestion here ? Am i supposed to use either krisp or BVCTelephony?tall-airline-16756
08/06/2025, 12:21 PMtall-airline-16756
08/06/2025, 12:22 PMkrisp_enabled
attribute in the CreateSIPParticipant
API.
• Verify that the Krisp integration is functioning as expected. It uses AI to remove background noise and should help improve call clarity.
Both Krisp and the BVCTelephony
agent code can be used together. They serve different purposes, and there is no requirement to choose one over the other. If the issue persists, it might be worth reviewing the audio setup or testing with different configurations to identify the root cause of the echo.
Sources: SIP overview | LiveKit Docs | SIP overview | LiveKit Docssalmon-leather-15461
08/06/2025, 12:26 PMtall-airline-16756
08/06/2025, 12:27 PMkrisp_enabled
, let’s try a few more troubleshooting steps together:
1. Double-check Krisp Setup:
◦ Make sure krisp_enabled
is correctly set up in both your inbound trunk and outbound call settings.
◦ Keep an eye on the SIP call status and enabled features to confirm Krisp is actually active during the call.
1. Audio and Media Settings:
◦ Tweak the jitter buffer settings to better handle your network conditions.
◦ Verify that your telephony codecs (like PCMU/PCMA) and WebRTC codecs (like Opus) are playing nicely together.
◦ Check your media encryption settings to make sure audio processing is smooth.
1. Network and Connectivity:
◦ Confirm that NAT traversal settings and external IP configurations are spot on.
◦ Make sure RTP media ports are accessible and properly configured.
1. Advanced Krisp Troubleshooting:
◦ Try enabling Background Voice Cancellation (BVC) to boost echo and noise reduction.
◦ Check that WebRTC echo cancellation settings aren’t conflicting with Krisp’s features.
If you’re still having trouble, feel free to share your current SIP trunk setup, dispatch rule settings, and any relevant logs or error messages. That way, we can dig in together and find exactly what’s causing the echo.
Sources: Noise & echo cancellation | LiveKit Docs | livekit-protocol/livekit/protocol/sip.pyi | SIP overview | LiveKit Docs | SIP Integration | livekit/livekit_composite | DeepWiki | Noise & echo cancellation | LiveKit Docssalmon-leather-15461
08/06/2025, 12:30 PMtall-airline-16756
08/06/2025, 12:30 PMCreateSIPParticipant
API.
If you're sticking with the default setup, Krisp might not be working on your inbound calls, which could be why you're hearing that echo. I’d suggest turning Krisp on in your SIP trunk settings and double-checking that it’s active for both inbound and outbound calls.
Need a hand figuring out how to tweak these settings? Just give me a shout—I’m happy to help!salmon-leather-15461
08/06/2025, 1:28 PMtall-airline-16756
08/06/2025, 1:28 PMmedia_use_external_ip
and media_nat_1_to_1_ip
, are set up correctly to avoid any network-related echo problems.
1. Tweak Your Audio Settings:
◦ Try enabling the jitter buffer—it helps smooth out any audio bumps.
◦ Check that your audio codecs are the best fit for your setup since some can cause more echo than others.
1. Double-Check Krisp’s Setup:
◦ Make sure Krisp is switched on for both incoming and outgoing calls.
1. Look Over Media Encryption:
◦ Have a peek at your media encryption settings to ensure they’re not getting in the way of Krisp’s noise cancellation.
If the echo is still being stubborn, feel free to share some details about your network setup, codec choices, and any logs you have. That way, we can dig in together and figure out what’s causing it.
Sources: livekit/livekit_sip.pb.go | Speech & Audio Processing | livekit/livekit_composite | DeepWiki | SIP Integration | livekit/livekit_composite | DeepWiki | SIP overview | LiveKit Docs | SIP APIs | LiveKit Docssalmon-leather-15461
08/06/2025, 1:38 PMtall-airline-16756
08/06/2025, 1:38 PMsip:
# Other SIP configuration parameters...
enable_jitter_buffer: true
This tweak helps smooth out those annoying network delays and makes your call audio sound better. If you have any questions or need a hand with this, just let me know—I’m here to help!salmon-leather-15461
08/06/2025, 1:40 PMtall-airline-16756
08/06/2025, 1:41 PMenable_jitter_buffer: true
in your SIP configuration file.
If you have any questions or want a hand setting this up, feel free to reach out—I’m here to help!salmon-leather-15461
08/06/2025, 1:41 PMtall-airline-16756
08/06/2025, 1:42 PMconfig.yaml
. Where you find it really depends on how you’ve set things up, but most of the time it’s:
• Mounted as a volume in your server or container.
• Set through an environment variable (like SIP_CONFIG_FILE
).
If you want to enable the jitter buffer, just pop this into your config.yaml
file:
enable_jitter_buffer: true
Not quite sure where that file is in your setup? No worries! Take a peek at your deployment docs or check your environment variables. And if you want, I’m here to help you find or tweak that file anytime.