SIP Advanced

Advanced SIP Settings

Far End Nat Traversal (FENT)    Requires license. Buy Evaluate.

- Keep-alive packets interval  UDP  seconds  TCP  seconds

(to keep SIP communications alive for clients needing FENT)

- Type of keep-alive packets  SIP REGISTER   SIP OPTIONS

Detect endpoints behind same NAT (for shortest media path)

Authorized Users

Method URI Direction Authentication User IDs

TLS settings Requires license. Buy Evaluate.

Act as TLS Client towards Server IP address(es)

Note: TLS is configured automatically on every network interface if any certificates has been installed in the unit. However if you want to override the default configuration, use the table below. The default configuration uses the first server certificate installed in the unit and all trusted certificates. Interop is enabled and MTLS is not.

Network interface Certificate Trusted certificates Server methods Client methods MTLS Interop.

SIP proxy

Maximum number of active sessions (Buy More)

Maximum number of registrations per user

Timeout before registration expires seconds

Call time out seconds

Enable SIP session timer

Default ring timeout seconds

Maximum ring timeout seconds

SIP Server UDP port numbers

SIP Server TCP port numbers

SIP Server TLS port numbers

Static domain forwarding

Domain Forward to Modify URI Authenticate

Allowed Codecs

Enable Codec Filtering

Codec Name Media

Codec Manipulation

Add (out-of-band) DTMF payload type

in SDP when sending to IP address(es)

Proxy rules

Proxy default access limit

IP Address Subnet Mask Access limit

Advanced

SIP (3xx) redirection handling

TCP connection timeout seconds

Max number of media streams per SIP request

Accept

Always record route

Music on Hold URI

RTP media port range -

Allow multiple RTP media senders

Allow RTP in reverse direction

Reuse port numbers within same session

Reuse port numbers when changing media (e.g. T.38 FAX)

Spoof protection

RFC 3261 Loose routing

Disable change to TCP (large messages)

Disable username encoding (registrations)

Allow RFC 2069 authentication

Preferred QOP (authentication)

URI Encoding

Try make UDP messages smaller when exceeding bytes

Hide Via headers on requests to IP addresses:

Ignore required extensions

Trusted IP addresses (treated as inside)

Monitor servers for failures

Time to cache server failures seconds

Client registration interval seconds

Failed registration retry timeout seconds

Disable media based forwarding

Find SIP URI locally

Strip ICE SDP attributes

B2BUA re-INVITE end-to-end

Anonymous calls (B2BUA)

Don't send error response (408) on timeout

Strip route information from 180 responses

Trusted networks (RFC 3325)

Enable

Default Privacy Policy

IP address range

Begin End Protocol

Handling of Call Transfer

Enable B2BUA to be able to transfer calls:

Always

For clients not supporting "REFER"

For clients not supporting "replaces"

For clients with From-URI:

Wildcards (*, ? and !) allowed (e.g. *@supercall.com john.brown@mysip.com)

For clients with User-Agent field:

Wildcards (*, ? and !) allowed (e.g. "Super Phone*" RTC/1.2)

B2BUA can also be enabled for outgoing calls through Dial Plan with the
b2bua parameter, e.g. Forward to: pstn.gw.com;b2bua

Forwarding to SIP account in dial plan will always use B2BUA

Quality of Service

Set DiffServ bits in all SIP media streams

      - audio

      - video

      - data

      - signalling

      - RTSP

      - TR-069

Strip DiffServ bits from other applications

Upstream prioritization (QoS)

Enable

      - Link capacity kbps

Internal Task Prioritization

Voice (media) quality vs call control

Call Admission Control

Deny calls if any of the following is exceeded:

- Upstream traffic kbps

- Downstream traffic kbps

- Audio calls

- Video calls

SIP Switch