(v12.0.8874.35714 started 2024-04-18 18:00:37 on VM2)
SIP Support: Acts as stateless SIP proxy and forwards requests to DestinationIP, usually DestinationPort 5060 will be configured. GatewayPort may be any port, but usually port 5060 will be configured, in combination with TLS/SSL port 5061 is used normally. Supports SIP requests/responses from clients via TCP, TLS (keyword SSL specified) and UDP. Outgoing SIP connections to SIP servers only via TCP. Acts as proxy for RTP and RTCP traffic (SDP requests/responses ("m=audio..." and IP addresses) will be modified accordingly). UDP ports used for RTP and RTCP at application gateway: ..client side: 50000-50499 ..server side: 60000-60499
Example for configuation of Domain Proxy within SIP client (e.g. the free SIP-based softphone X-Lite from Counterpath): GatewayIP:GatewayPort;transport=tcp
Successfuly running since years using the voice over IP provider A1.
Sub-switch IGNORE:method ... method is a list of SIP methods separated by | e.g. NOTIFY|PUBLISH|SUBSCRIBE - These methods are ignored and not forwarded. Useful if server does not support these methods and client does not stop sending these requests. Sub-switch LLP:option ... Local Loop Processing, see also following Configuration overview - Local loop processing when calling SIP (softphone) address from internal client, option is optional - If LLP:NUM is specified: phone calls also via local loops if telephone numbers match (phone number must start with "+" or "00") - Warning: If cascaded Application Gateways are used: NUM must be specified at the first SIP proxy (from the view of the client) if any upstream proxy has specified the NUM option. Otherwise loops will not work. - If called client has a SIP session via this proxy, the call will be routed directly to the client bypassing the SIP server. - If SIP sesions are routed via the same GatewayIP, voice and video traffic (RTP) is now routed directly between the clients (peer-to-peer). - Both clients (caller and called party) must have the LLP switch specified. Otherwise call setup and RTP traffic will be routed via the SIP server. Sub-switch NALT: - Removes any a=alt SDP commands to force communication via primary address of SIP server. Sub-switch NOSENC: ... No SIP encryption - Removes any "a=crypto" in SDP section of SIP command INVITE to server - Allows compatibility with SIP servers and clients that do not support encryption but client offers encryption. - Additionally SIP proxy monitors if first hop of client is via TLS. If not: SIP proxy removes "a=crypto" for loops. Sub-switch NRR: - experimental feature: new Record-Route processing, not compatible with sip.a1.net!. Sub-switch PRTP: - When LLP is specified: RTP traffic is routed via this proxy (but not via the SIP server) even if GatewayIP of both SIP sessions is identical. Useful if clients are behind NAT devices. Sub-switch UDP: - SIP clients access proxy via UDP. Default: TCP, outgoing connections are TCP in any case. - When using UDP with SIP: GatewayIP must be defined explicitly ("*" is not supported). - Note: Reduce TTL to expiration (see response to REGISTER, Contact: ... ;expires=nn ... nn are number of seconds Sub-switch VID:n : - Allows video streams. Default: No video streams allowed. - n specifies the size of the receive buffer to be used for the RTP video stream, n is optional, 8000 <= n <= 100000, default is 32000. Firewall settings for A1overIP (Status: 2010-01-01): outgoing to 80.75.55.109 (sip.a1.net) TCP 5060, 80.75.55.93 UDP 50000-50500
For status of SIP sessions: LGC:v to log phone call duration, v is optional and may be 0,1,2 or 3 (default is 0), details see here. RGS Valid only if GatewayIP2=status: Output of list of registered SIP users only, dependent on string in URL: string ... list following: sip ...... all SIP sessions with sip URI tel ...... all SIP sessions with tel URI enc ...... encrypted SIP sessions only noe ...... not encryted SIP sessions only .. e.g. telenc lists encrypted SIP sessions with tel URI .. if URL does not contain tel and sip and does not contain enc and noe: list all SIP sessions with sip URI Sub-switches for RGS (optional): .. SIZE:size ... font size in px, default is 16 .. WIDTH:width ... table width in px, default is 300 RSL Valid only if GatewayIP2=manage or status: Allows reading of current SIP logfile via URL SL.csv .
