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Help fix OpenSIPS configuration for SIPPY B2BUA

I’m looking for OpenSIPS expert to help me create proper configuration file for use with Sippy B2BUA.

Scenario is really simple, but configuration files that I found as examples on the net doesn’t work 100% as expected. Tried to use configuration from the official OpenSIPS book but with no success. There are issues with NAT’ed BYE requests and with ACK’s, so some modification is needed.

Current config file is: [url removed, login to view]

Design goals:

1. OpenSIPS version 1.72 notls as registrar (MySQL)

2. NAT support for all popular NAT types, as much compatible with various softphones/devices as possible. Support for both NAT’ed and not NAT’ed clients. Also support for devices behind NAT but with public IP addresses in SIP requests (changed on client side by using STUN, ICE etc)

3. All inbound/outbound calls after authentication should be passed to Sippy B2BUA and it will handle routing logic and establish connections with B-leg of the call

4. Sippy is running on the same host but different port: 5061

5. Sippy is firewalled and it will accept connections only from OpenSIPS IP address. No direct connections with clients. Because of this OpenSIPS must stay in the middle of all SIP dialogs and properly forward all types of requests up and downstream

6. Inbound calls from SIP provider should not be challenged for password. IP based authorization only

7. Sippy will try to detect idle connections (ie lost BYE packet) by making re-invite and it also needs to be handled

8. Only Sippy is responsible for rewriting SDP and deciding to use or not use RtpProxy. No RtpProxy handling on OpenSIPS side is needed

Desired call flows:

Outbound call (remote SIP client) -> OpenSIPS -> Sippy -> SIP provider

Inbound call (SIP provider) -> OpenSIPS -> Sippy -> remote SIP client (user registered to OpenSIPS)

Detailed issues with current config:

1. 404 Not Here error for BYE and CANCEL requests. Has_totag returns true but loose_route() returns false. Don’t know why

2. To solve above problem I add if (is_method("BYE") && $si != '[url removed, login to view]') { send("udp:[url removed, login to view]"); exit; } before send_reply("404","Not here");

It helps with BYE, but I’m not sure is it proper way of dealing with this problem

3. I see Sippy address/port in Contact of some replies generated by OpenSIPS. It should be replaced with OpenSIPS address/port. Potential re-invites from clients should go through OpenSIPS not directly to Sippy since it’s firewalled

4. When outbound call is established OpenSIPS does not forward OK to the client and call is dropped after 30 second timeout

So I’m looking for help in either correcting my current [url removed, login to view] file or for a complete new configuration file that will work in my scenario. Also if you have any field-proven arguments against this configuration or think that something should be done in other way I’m open to any suggestions.

Your work will be considered done when provided [url removed, login to view] file will make my OpenSIPS server working according to design goals listed above.

Kĩ năng: Linux, VoIP

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( 5 nhận xét ) Warszawa, Poland

ID dự án: #1644073

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Hello Adam, I can do this job as we discussed. Thx. Andrew

$120 USD trong 2 ngày
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can give you config for kamailio latest. it work ok with asterisk in same config. can be tunned to ur system. any fixes if needed included. for old version of kamailio and opensips this is not fixable.

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