Documentation
Connect PortaSwitch to Open Voice Shield
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In PortaSwitch, add Open Voice Shield as a vendor with a Calls to vendor via SIP connection to sip.ovs.telecomsxchange.com port 5060, IP-based, G.711 first; attach a vendor tariff and include the routes in a routing plan. Whitelist your PortaSIP node's public IP on the Network page first.
Before you start#
- An Open Voice Shield account. Signup is self-serve and starts you with US$5 of credit.
- The public IP address your PortaSIP node sends SIP from, the address after any NAT, not the one on its interface.
- A destination, the carrier the calls were already going to. We relay each accepted call on to it.
- Firewall: UDP or TCP 5060 to the ingress, and UDP 30000–40000 open in both directions for the RTP your media endpoints exchange with our relay. See the firewall rules.
Step 1: whitelist your PortaSIP node's IP#
Open the Network page in the portal and add the public address your PortaSIP node sends from as a CIDR entry (a single host is a /32). The whitelist is the authentication: there are no credentials, and an INVITE from any other address is answered 403.
Step 2: define the vendor and its SIP connection#
PortaSwitch reaches a carrier through a vendor and its connection, and routes exist only for destinations a vendor tariff rates. Create a vendor for Open Voice Shield, add a Calls to vendor via SIP connection, and attach a tariff rating the destinations you will send. The rates you enter there are your own cost records.
| Field | Value |
|---|---|
| Remote gateway | the field naming the far-end SIP peer: sip.ovs.telecomsxchange.com, confirm the address on your Network page |
| Port | 5060 |
| Transport | UDP or TCP |
| Authentication | none: no login or password; authorization is your whitelisted source IP |
| Codecs | G.711 (PCMU, PCMA) first, as plain RTP, no SRTP towards the ingress |
| Registration | none: a REGISTER at the ingress is answered 403 |
Screen names follow PortaOne documentation conventions. Menus vary by model and firmware, so consult your version's manual.
- The INVITE must carry SDP, an early offer. An INVITE without SDP is answered 488.
- Offer G.711 (PCMU or PCMA) first; G.722, G.729 and G.723.1 are decoded as well. Any other codec is relayed but produces no transcript and no verdict.
- Send no REGISTER at the ingress: it is answered 403. Calls are authorized by source IP alone.
Step 3: include it in a routing plan#
The connection's routes appear once the tariff rates a destination. In the routing plan your traffic uses, give them a preference: above your direct carrier to send everything through the analyzed hop, or alongside it to share the load. Open Voice Shield relays each accepted call on to your configured destination, so the analyzed route is a complete route: the call reaches the same place, one hop longer. Keep your direct carrier in the plan as the failover.
Or route a rated subset#
You do not have to move all your traffic: sampling is a supported pattern. Where your release supports weighted or percentage distribution among routes of equal preference, give the analyzed route a share (one call in ten is a common start) and your direct carrier the rest. Otherwise sample deterministically with a subset: rate only a prefix range in the vendor tariff, for example only destinations beginning 44161, so that only calls to Manchester take the analyzed hop, and every other destination keeps its direct route.
Both paths deliver the call: the analyzed share reaches the same destination through one extra hop, and every sampled call comes back with a recording, per-channel transcript and verdict.
Step 4: place a test call#
Send a real call through the new route and keep both parties talking for at least 30 seconds. A leg that never speaks produces a one-sided transcript. After hang-up the call appears in Calls in the console with its recording, per-channel transcript and verdict, usually within a minute.
- Rejected 403? The source address is not whitelisted. Compare your own SIP trace with the CIDR entries on the Network page.
- No route offered? The vendor tariff does not rate the dialled destination, or the connection is not in the routing plan the call used.
- One-way audio? A firewall rule on UDP 30000–40000 between your media endpoints and the ingress, not a signalling problem.
Anything else: Troubleshooting lists every rejection code and the check that clears it.
Frequently asked questions#
Does PortaSwitch register to Open Voice Shield?
No. Nothing registers in either direction: a REGISTER sent to the ingress is answered 403, and calls are authorized by the source IP you whitelist on the Network page, so the connection carries no credentials.
Why does the connection need a vendor tariff?
Because that is PortaSwitch's routing model: a vendor connection produces routes only for the destinations its tariff rates, and a routing plan can only order routes that exist. The rates you enter are your own cost records. Open Voice Shield bills analyzed minutes separately, per its pricing page.
Can I send only some destinations through it?
Yes. Rate only the prefixes you want analyzed in the vendor tariff attached to the connection. The routing plan then offers the analyzed hop for those destinations alone, and everything else keeps its direct route.
Ready to try it? Create an account. Signup is self-serve and comes with US$5 of credit, enough to analyze your first test calls.