> ## Documentation Index > Fetch the complete documentation index at: https://docs.invoca.com/llms.txt > Use this file to discover all available pages before exploring further. # SIP Integration: Specifications & FAQs > Advanced technical specifications and frequently asked questions for troubleshooting your Invoca SIP integration. After connecting your Invoca calls to your business over the internet using Session Initiation Protocol (SIP), you might have more questions. This article covers advanced technical specifications and frequently asked questions about your SIP integration. ## Advanced Invoca SIP technical specifications * Traffic is routed over public IPv4 addresses for both SIP and RTP. * DTMF is always transmitted using RFC 2833; in-band DTMF isn't supported. * The codec is always G.711μ (ulaw). * DNIS (access number) and ANI (Caller ID) phone numbers default to E.164 format, optionally preceded by `+`. * Invoca doesn't support SIP authentication or the T.38 fax protocol. * Invoca uses the Session Border Controller (SBC) type Kamailio v5.5.3. * When SIP is integrated, all other Invoca platform features operate normally. * Your call center's data provider must have direct peering with Amazon Web Services (AWS) and Google. * An MPLS or VPN generally isn't necessary, though you may use a commercial product like AT\&T NetBond if you prefer. * Invoca uses wholesale SIP connectivity to Tier 1 carriers in the US, and directly to customers — 100% SIP connections with no translation to SS7, so there's no risk of losing SIP headers or audio degradation from transcoding. ## FAQs **Does SIP run over the public internet?** Yes — but Invoca runs on Amazon and Google's cloud, which have direct peering with virtually everyone. In practice, running a traceroute typically shows no public internet between Invoca's platform and yours; the connection is effectively peer-to-peer. **Is there a risk the SIP integration will affect audio quality, or require a dedicated network or QoS protocol?** No. Those are real risks in corporate and home environments, but not on the backbone networks Amazon and Google run, where Invoca's platform operates. **How do I see previous changes to my SIP integration?** Use your account's change history, then filter by type: SIP Integration. **How do I check the status of my SIP integration?** Check the Overview tile on your SIP settings page — select the Profile used to install your SIP integration, hover over the Integrations menu, and select SIP Integration. This tile shows the integration's status and when settings were last modified, including changes to SIP endpoints, phone number format, and SIP headers, and when Verify was last used to test endpoints. Hovering over the Active or Deactivated status shows who activated or deactivated it, and when. **What does a sample SIP INVITE from Invoca look like?** ``` INVITE sip:18555550146@10.10.19.195 SIP/2.0 Via: SIP/2.0/UDP 18.206.253.88;rport;branch=z9hG4bK9e03gy70Uge1Q Max-Forwards: 69 From: ;tag=4NU6QKtyrNNHj To: Call-ID: 5144a4a8-0aef-1237-ee9d-aaee0f6e5e0d CSeq: 125939874 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.19+git-20171030T201728Z~3a73c4e56f~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 242 X-InvocaID: 93B6-8E5BECCFAAAA Remote-Party-ID: ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1532531871 1532531872 IN IP4 10.10.19.67 s=FreeSWITCH c=IN IP4 10.10.19.67 t=0 0 m=audio 20006 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 ``` You can use this sample to troubleshoot your SIP configuration and prepare your destination phone systems for your Invoca SIP integration. ## Where to go next * [SIP Integration Overview](/s/article/jhdcp92865-601)