SIP Trunking - How it works

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  • To request a SIP trunk interop with Thinktel, please fill our this form

The following are the basic setting required to start a SIP Trunk Interop with JumpShop's network.

  1. Ensure your gateway supports DNS SRV records.
  2. Allow the following codecs: g.711, g.726-32 / g.721, g.729 & T.38 (Fax over IP). Asterisk, Trixbox, OpenPBX peers cannot use g.726-32 / g.721.
  3. Ensure you allow our proxies to contact you:
    • SIP, UDP port 5060: 208.68.17.144 (eico.thinktel.ca) if your PBX is on a routable IP, 159.18.161.67 (vp.thinktel.ca) if your PBX is behind NAT. 
    • RTP, UDP ports 10000-65500 from the following IPs: 159.18.161.99, 159.18.161.104 to 159.18.161.109 (total 6 in range), 208.68.17.40,  208.68.17.51 to 208.68.17.53, 206.80.250.8 to 206.80.250.19, 208.68.18.67
  4. Configure your gateway to send and receive 10 digit North American numbers. 11 digit is accepted as well.
  5. Configure DTMF for RFC2833 (out of band)
  6. Every 30s our switch sends out a SIP 'keep alive' for your trunk which is essentially a SIP options request, to which we expect any SIP reply. If we receive no response within 30s of the SIP options request, the switch turns the SIP trunk state to disabled and will not send further SIP invites until the keep alive gets a response by aggresively sending out options requests every 4 seconds.
  7. We authenticate calls by the IP address in the SIP Contact header and username and password.
  8. The SIP Contact IP & port in the header must match what the IP and port defined in our switch.
  9. If you are running Asterisk, TrixBox or OpenPBX, please ensure that canreinvite=no is set on the SIP settings. Version 1.2.x and earlier may also experience DTMF issues. It is recommended to use a newer version if possible. Otherwise, check the [Asterisk 1.2 DTMF Patch] for more details.
  10. We allow up to 8 IP addresses per SIP trunk in our switch. You are allowed to use an IP address across seperate SIP trunks if the SIP service port is unique on each SIP trunk. We identify each SIP trunk with a pilot number, which is a 10 digit number local to your VoIP gateway, similar to how Telco's assign pilot numbers to conventional PRI's. Please refer to your pilot number when placing orders for DIDs,channels,etc. The requested number of channels in a SIP trunk activated once the intial interop stage is complete and your account is active with JumpShop / Thinktel.
  11. Each IP in the trunk is used for either origination or termination to our switch. It is really up to you which IPs are used for what, to us it is transparent. With calls terminating to you (on DIDs we have assinged to your SIP Trunk), we send the SIP invite in a round-robin fashion to each IP address/port defined in your SIP trunk. If your SIP response is > 303 (404/Not Found or 503/Unavailable the proper response), we try the next IP address until the Invite is accepted with a SIP response < 303. T
  12. A single contact IP address cannot be used with multiple trunks with the exception of utilizing an alternate contact port. If you are ordering an additional trunk please submit a unique IP differing than any currently in use with our service. Or an existing IP with alternate contact port (other than udp 5060).

PLEASE NOTE:

* If you're behind NAT and your router has SIP ALG enabled, please disable this as the option is NOT compatible with our service. If enabled you may experience registration or audio related problems  potentially others that will negatively impact your ability to use our service. *

Retrieved from "http://wiki.jumpshop.com/SIP_Trunking_-_How_it_works"
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