Models with a V in their name contain a VoIP/SIP client. Some models even contain sRTP. However, DrayTek decided to exchange the keys for sRTP not via SDES but ZRTP. Therefore, DrayTek does not need SIP-over-TLS and SIP-over-UDP is sufficient. Although ZRTP has many benefits, major SIP B2BUA like Digium Asterisk destroy it. However, if your callee offers a SIP-URI, you are able to dial him directly and therefore leverage ZRTP. For this, do not specify the SIP Proxy as Outbound Proxy or go for VoIP → General → IP Call. DrayTek Taiwan calls this Peer-to-Peer Calling. DrayTek UK calls this SIP-to-SIP Calling. I call this SIP-URI Dialling.
Password: | admin/admin System Maintenance → Administrator Password |
HTTPS: | enabled on default System Maintenance → Management → (tab) IP version → LAN Access Control → Enforce HTTPS Access |
Update: | System Maintenance → Firmware Upgrade The letters after the current firmware version (also printed on the sales box) indicate the modem-code variant: For example, VT1 means VECTOR1. Furthermore, your local distributor might offer a localized firmware: For example, for German you have to go for the German distributor. |
SIP-URI User: | VoIP → SIP Accounts → Index → Account Number/Name |
SIP-URI Host: | VoIP → SIP Accounts → Index → Proxy … Domain/Realm … Register via: Auto … Ring Port: tick the desired Phone Port(s) |
ZRTP: | A) VoIP → General: Secure Phone B) VoIP → DialPlan → Phone Book → Index → Secure Phone: ZRTP |
Session Timers: | VoIP → Phone → Index → Session Timer: 1800 |
SHA-2 Digest: | does not pick MD5, continues without header Authorization, therefore is not able to register; therefore incompatible with Linphone |
DNS-SRV: | not enabled on default Mitigation via Command Line Interface (CLI): voip sip acc n -s 1 (n is the account 1 to 6) |
Audio DiffServ: | not enabled on default Mitigation: VoIP → General → RTP TOS: EF Class |
12 V 1.4 A, Coaxial: 5.5 mm × 2.1 mm