VoIP: SIP-over-TLS and sRTP: Grandstream

Grandstream offers a vast range of phones, asks for small prices, provides big data-sheets, and what about the software? Let us have a look! By the way, their Webpage is using Mixed Content. Although two years after reporting this, it is not fixed as of today.

Last tested firmware
retested in Oct. 2019 with
retested in May 2020 with (DP750) and (GRP2612)


Password: admin/sticker on back or admin
Web → Maintenance → Web Access → Admin Password
HTTPS: Web → Maintenance → Security → Web → Access Mode: HTTPS
Update: Web → Maintenance → Firmware → Server Path: firmware.grandstream.com → Button: Save and Apply → Status → System info: Check for New Firmware
the factory/default path does not work; you have to change this first
Trust Anchors: Web → Profile or Account → Security
1. Authenticate Server Certificate Domain: Yes (broken in GRP2612)
2. Authenticate Server Certificate Chain: Yes (broken in GRP2612 if used with Custom Trusted CA Certificates)
3a. (DP750) Trusted CA Certificate: Base64
you have to paste the Base64 of your trust anchor; only one certificate possible; you have to save, apply, and reboot; you do not see your certificate
3b. (GRP2612) Web → Maintenance → Security → Trusted CA Certificates: Base64 → Load: Custom
because you cannot double-check the Default Trusted CA Certificates; currently broken, instead you have to use ‘All’
SIP-URI User: Web → (DECT → SIP) Account → SIP User ID
SIP-URI Host: Web → Profile or Account → General → SIP Server
SIP-over-TLS: Web → Profile or Account → Network → DNS Mode: NAPTR
Web → Profile or Account → SIP → Basic → Transport: TLS
Web → Profile or Account → SIP → Basic → Use Random SIP Port: Yes
SDES-sRTP: Web → Profile or Account → Audio → Crypto Life Time: No
Web → Profile or Account → Audio → SRTP Mode: Enabled But Not Forced
which is RTP/SAVP; Optional is RTP/AVP + RTP/SAVP = sRTP is second; therefore Digium Asterisk (and DUStel which use Asterisk) do not go for sRTP

Bad Defaults

These features are disabled on default, although they are automatically negotiated. Tests revealed they work. Therefore, no reason exists to disable those on default.

Session Timers: Web → Profile or Account → SIP Settings → Session Timers
1. Enable Session Timers: Yes
2. Session Expiration: 1800
3. UAC Specify Refresher: Omit
PRACK: Web → Profile or Account → SIP → Basic → 100rel: Yes

Software Bugs

SHA-2 Digest: crashes when several digests exist; therefore incompatible with Linphone
DNS-NAPTR: has not effect in the GRP2612
AES-256 sRTP: wrong name and the key is not 46 but 48 bytes long
Audio: G.726-32 has the wrong endianness on default and ‘G726-32 Packing Mode’ has no effect
Mitigation: set ‘g726nonstandard=yes’ in your Digium Asterisk
Opus-Codec gives just buzzing noise
Mitigation: set the ‘Preferred Vocoder’ 6 and 8 to the same as 1; this disables Opus-Codec and G.721 (aka G.726-32).
Compact Form: Supported (k) and Session-Expires (x) are not understood
Mitigation: Web → Profile or Account → SIP → Session Timers → Enable Session Timers: No


Bugs: padlock icon even without authenticated transport
Cipher Suites include RC4 (even MD5)
Privacy: phones home to https://acs.gdms.cloud
Mitigation: Web → Maintenance → TR-069: ACS URL: empty
phones home to http://fm.grandstream.com/gs
Mitigation: Web → Maintenance → Upgrade → Firmware Server empty
Mitigation: Web → Maintenance → Provisioning → Config Server: empty
phones home to https://service.ipvideotalk.com
Mitigation: Web → Settings → Web Service → Auto Location: No
Responsible Disclosure: although a ticket system exists, it does not work:
I had to go to a venue and get the business card of a Product Manager.
After an issue gets fixed, the corresponding ticket is not updated/closed.
Instead, you have to go through the Release Notes, whether your issue got fixed.
Firmware Update: Newsletter


Model Range (non-Android)

Power Supply

DP750: 5 V 1 A, Micro-USB
GRP2612: 5 V 0.6 A, Coaxial: 5.5 mm × 2.1 mm

back to the other phones.