Re: [Kamailio-Users] Advanced Call Scenario / One Way Audio
Am 10.03.2010 21:33, schrieb Brandon Armstead: REGISTRAR-01 AND REGISTRAR-02 are both proxying RTP As well as the initial Asterisk in the middle SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Having 3 media relays is a bit strange. Only one should be enough (e.g. Asterisk). Use a packet sniffer and verify who is sending RTP packets, and where the RTP flow stop. Then analyze the SDPs seen by the component where the RTP stream stops. regards klaus ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] Advanced Call Scenario / One Way Audio
**resending as I am not sure if this made it out the first time, I do not believe it did, if this is a duplicate -- my apologies.** Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 - REGISTRAR-01 - Kamailio-01 - Asterisk (New Call-ID + Asterisk in Media Path) - Kamailio-01 - REGISTRAR-02 - UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared USRLOC database to simulate calls from PSTN to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner traversal This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 - UA2, and forces P2 - UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this bridge / interconnection We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both proxying RTP As well as the initial Asterisk in the middle SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
[Kamailio-Users] Advanced Call Scenario / One Way Audio
Hello, As always, thank you for all / any help and input you may provide in advance. Call Scenario: UA1 - REGISTRAR-01 - Kamailio-01 - Asterisk (New Call-ID + Asterisk in Media Path) - Kamailio-01 - REGISTRAR-02 - UA2 UA1 is behind NAT UA2 is behind NAT The purpose of this is when using a shared USRLOC database to simulate calls from PSTN to generate both legs of the call, i.e. incoming and outgoing, and also allow for easier / cleaner traversal This aids from scenario's happening where UA1 calls UA2 (while UA1 exists on P1 and UA2 exists on P2) this prevents P1 - UA2, and forces P2 - UA2 We determine that this is a call from P1 to P2 (internal call) and thus create this bridge / interconnection We are running into a problem it seems with one way audio, i.e. the CALLEE can hear the CALLER, however the CALLER CAN NOT hear the CALLEE. REGISTRAR-01 AND REGISTRAR-02 are both proxying RTP As well as the initial Asterisk in the middle SDP. Let me know if this makes sense and if you guys have any further thoughts on what may possibily be going wrong. Perhaps there are better ways to go about this, let me know if I am way off course, thank you! Sincerely, Brandon Armstead ___ Kamailio (OpenSER) - Users mailing list Users@lists.kamailio.org http://lists.kamailio.org/cgi-bin/mailman/listinfo/users http://lists.openser-project.org/cgi-bin/mailman/listinfo/users
