@millions-hair-24956 is it possible to use RCD with sip service?

The data sources do not explicitly mention RCD (Ringback Call Diversion) in relation to LiveKit's SIP service. However, LiveKit's SIP service does support various SIP functionalities, including inbound and outbound call handling, SIP trunk configuration, and participant management, as detailed in the SIP Overview and SIP APIs documentation. If you are referring to a specific feature or functionality related to RCD, could you provide more details or clarify your requirements? This will help in determining if LiveKit's SIP service can meet your needs or if there are alternative solutions. Sources: - SIP Overview - SIP APIs 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 i was talking about Rich Call Data (RCD)

I'm not able to find any data sources to answer your question. You might want to specify how you intend to integrate Rich Call Data (RCD) with LiveKit's SIP service. If this doesn't help, please reach out to the Livekit support team for further assistance. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 is it possible to edit the stir/shaken header for an SIP call?

It seems likely that editing the STIR/SHAKEN header directly in LiveKit's SIP service is not explicitly mentioned in the available data sources. However, LiveKit does support custom SIP headers through the

headers_to_attributes

configuration, which allows mapping of

X-*

headers to participant attributes, as noted in the SIP participant documentation. If you are looking to manipulate specific headers like STIR/SHAKEN, you might need to handle this through your SIP provider or a proxy server that can modify SIP headers before they reach LiveKit. For more detailed guidance, you may want to consult with your SIP provider or explore using a SIP proxy like Kamailio or FreeSWITCH. Would you like more detailed information on configuring custom headers or using a proxy server? Sources: - SIP participant documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 My use case is for outbound calling, i am using twilio and i want to add data in my sip STIR/SHAKEN header

It seems likely that directly modifying the STIR/SHAKEN header for outbound calls using Twilio with LiveKit is not explicitly supported in the available data sources. LiveKit allows for custom SIP headers through the

headers_to_attributes

configuration, but this typically applies to

X-*

headers, as mentioned in the SIP participant documentation. For STIR/SHAKEN headers, which are used for caller authentication and verification, you would generally need to configure this through your SIP provider, in this case, Twilio. Twilio manages STIR/SHAKEN compliance and may provide options to include or modify these headers as part of their service. You might need to consult Twilio's documentation or support for specific capabilities related to STIR/SHAKEN. Would you like more detailed information on configuring custom headers or guidance on working with Twilio for STIR/SHAKEN? Sources: - SIP participant documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 Yes please, can you guide me for such an implementation?

To implement STIR/SHAKEN headers for outbound calls using Twilio with LiveKit, you would typically need to work directly with Twilio's capabilities for handling these headers, as LiveKit itself does not provide direct support for modifying STIR/SHAKEN headers. Here's a general approach you might consider: 1. Consult Twilio Documentation: Review Twilio's documentation on STIR/SHAKEN to understand how they handle these headers. Twilio is responsible for ensuring that calls are compliant with STIR/SHAKEN standards, and they may offer options to include or modify these headers as part of their service. 2. Twilio Support: Reach out to Twilio support for specific guidance on how to include additional data in the STIR/SHAKEN headers. They can provide detailed instructions or configurations that are specific to their platform. 3. Custom SIP Headers: If you need to include additional information in the SIP headers, you can use LiveKit's

headers_to_attributes

feature to map custom

X-*

headers to participant attributes. This is useful for passing additional metadata, but it may not directly affect STIR/SHAKEN headers. 4. SIP Proxy: Consider using a SIP proxy like Kamailio or FreeSWITCH if you need to manipulate SIP headers before they reach Twilio. A proxy can provide more control over the SIP messages and allow for custom header manipulation. 5. Testing and Validation: Once configured, thoroughly test your setup to ensure that the STIR/SHAKEN headers are being applied correctly and that calls are compliant with the required standards. If you need more detailed steps or specific configurations, I recommend consulting Twilio's support or documentation, as they will have the most accurate and up-to-date information regarding STIR/SHAKEN implementation. Sources: - https://livekit-users.slack.com/archives/C07FRP66XGE/p1730253168.794269 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 This is from twilio documentaion: For Elastic SIP Trunking customers, there is a new header called

X-Twilio-VerStat

, and a new

Identity

header with the SHAKEN PASSporT. When your application receives a request webhook that has a

StirVerstat

parameter, Twilio will implicitly pass the

StirVerstat

to the Client when you

<Dial><Client>

. The information in the

StirVerstat

parameter can be used to display a trust indicator to the recipient when an incoming call from the public telephone network has been verified under the SHAKEN/STIR framework.

I'm not able to find any data sources to answer your question. You might want to specify how you plan to use the

StirVerstat

parameter or the

X-Twilio-VerStat

and

Identity

headers in your application. If this doesn't help, please reach out to the Livekit support team for further assistance. 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 Can i customize the INVITE request to my sip provider?

Customizing the INVITE request to your SIP provider in LiveKit is possible to some extent by using custom SIP headers. According to the SIP participant documentation, you can use the

headers_to_attributes

configuration to map custom

X-*

headers to participant attributes. This allows you to include additional information in the SIP INVITE request. However, it's important to note that this customization is limited to

X-*

headers. If you need to modify other parts of the INVITE request, such as the STIR/SHAKEN headers, you might need to use a SIP proxy like Kamailio or FreeSWITCH to manipulate the SIP messages before they reach your provider. Would you like more detailed guidance on setting up custom headers or using a SIP proxy? Sources: - SIP participant documentation 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 Can you explain to me how to setup an sip proxy?

Setting up a SIP proxy involves several steps, and using a popular SIP proxy like Kamailio or FreeSWITCH can provide the flexibility needed to manipulate SIP messages. Here's a general guide to setting up a SIP proxy: 1. Choose a SIP Proxy: Decide whether to use Kamailio, FreeSWITCH, or another SIP proxy. Kamailio is known for its high performance and flexibility, while FreeSWITCH offers extensive features for both SIP and media handling. 2. Install the SIP Proxy: - Kamailio: Follow the official installation guide for your operating system. Typically, this involves adding the Kamailio repository and installing via package manager. - FreeSWITCH: Follow the FreeSWITCH installation guide for your platform. 3. Configure the SIP Proxy: - Kamailio: Edit the

kamailio.cfg

file to define your SIP routing logic. This includes setting up SIP domains, user authentication, and any custom header manipulation. - FreeSWITCH: Use the

freeswitch.xml

configuration file to set up SIP profiles, dial plans, and any custom SIP handling. 4. Set Up Networking: Ensure that your SIP proxy is accessible over the network. Configure firewall rules to allow SIP traffic (typically on port 5060 for UDP/TCP). 5. Test the Configuration: Use SIP clients or tools like

sngrep

to test SIP message flow through your proxy. Verify that the proxy is correctly handling and routing SIP messages. 6. Integrate with LiveKit: Point your LiveKit SIP service to the SIP proxy by configuring the SIP trunk to route through the proxy. This allows you to manipulate SIP messages before they reach the final destination. 7. Monitor and Maintain: Regularly monitor the SIP proxy for performance and security. Use logging and monitoring tools to track SIP traffic and troubleshoot issues. If you need more detailed steps or specific configurations, I recommend consulting the documentation for Kamailio or FreeSWITCH, as they provide comprehensive guides and examples. Would you like more detailed information on a specific SIP proxy or configuration? Sources: - https://livekit-users.slack.com/archives/C07FRP66XGE/p1732523840.584939 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

@millions-hair-24956 will this add delay to my infrastructure?

It seems likely that adding a SIP proxy to your infrastructure could introduce some additional latency, but the impact is generally minimal if the proxy is configured and optimized correctly. The latency introduced by a SIP proxy is typically due to the additional processing required for routing and any custom logic applied to SIP messages. To minimize latency, consider the following: 1. Optimize Configuration: Ensure that the SIP proxy is configured efficiently, with minimal processing overhead for each SIP message. 2. Geographical Proximity: Place the SIP proxy geographically close to your other infrastructure components to reduce network latency. 3. Resource Allocation: Allocate sufficient CPU and memory resources to the SIP proxy to handle the expected load without bottlenecks. 4. Network Optimization: Use a high-performance network setup with low-latency connections between the SIP proxy and other components. 5. Monitoring and Tuning: Continuously monitor the performance of the SIP proxy and make adjustments as needed to optimize for latency. Would you like more detailed guidance on optimizing a SIP proxy for low latency? Sources: - https://livekit-users.slack.com/archives/C07FRP66XGE/p1732874965.633319 - https://livekit-users.slack.com/archives/C048FRL1N2C/p1731141029.045569 👍 0 button 👎 0 button 💡 Hint: Mention @runllm in the thread for followups.

How did you go with this @gray-beach-22915?

Hi I am still in research phase for now, so I don’t…

I am researching into Kamailio, I will let you know how i go