I use an audio stream relayed by a client that has no audio DSP card. so when I stop to talk, no UDP audio packet is sent to FS. Do you know a trick to send fake UDP packet to keep the call alive ? (I need to keep rtp-timeout short to hang up leg B when leg A hangs up) Thanks F
For now it's the only way I found to make leg B hanged up. Apparently the Bye message is not received by the leg B when leg B when it is registered to another FS in a shared DB ----- Original Message ----- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Thursday, March 25, 2010 2:24 PM Subject: Re: [Freeswitch-users] send fake UDP packets This statement makes NO sense. RTP-Timeout isn't used to hangup the call the Bye message is... Can you elaborate on this a bit more? /b On Mar 25, 2010, at 1:20 PM, Madovsky wrote: (I need to keep rtp-timeout short to hang up leg B when leg A hangs up) ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
FS where leg A is registered receives well Bye message... ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:48.472225: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:49.473166: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:51.474108: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:55.475066: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:33:59.476206: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:03.477236: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:07.478224: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:11.479164: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ recv 408 bytes from udp/[10.10.10.138]:64098 at 18:34:15.480076: ------------------------------------------------------------------------ BYE sip:99999999999@10.10.10.138:5080;transport=udp SIP/2.0 Content-Length: 0 Via: SIP/2.0/UDP 10.10.10.138:64098;rport;branch=z9hG4bKvIdAhnrmr220KEV8KGrv6g.. From: "Carl Zenmaier" <sip:8511234567890@mydomain.com>;tag=50140543427 To: <sip:99999999999@mydomain.com>;tag=Nt8UN6erUFK3p Contact: <sip:8511234567890@10.10.10.138:64098> CSeq: 3 BYE Max-Forwards: 70 Call-ID: 1044306705@10.10.10.138 ------------------------------------------------------------------------ but leg B doesn't receive anything.... I don't know yet if it's a client or FS problem. try to figure it out ----- Original Message ----- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Thursday, March 25, 2010 2:24 PM Subject: Re: [Freeswitch-users] send fake UDP packets This statement makes NO sense. RTP-Timeout isn't used to hangup the call the Bye message is... Can you elaborate on this a bit more? /b On Mar 25, 2010, at 1:20 PM, Madovsky wrote: (I need to keep rtp-timeout short to hang up leg B when leg A hangs up) ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
So apparently the leg B doens't receive any Bye message... but FS receives Bye message well from legA. maybe I missed a config.. Thanks ----- Original Message ----- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Thursday, March 25, 2010 2:24 PM Subject: Re: [Freeswitch-users] send fake UDP packets This statement makes NO sense. RTP-Timeout isn't used to hangup the call the Bye message is... Can you elaborate on this a bit more? /b On Mar 25, 2010, at 1:20 PM, Madovsky wrote: (I need to keep rtp-timeout short to hang up leg B when leg A hangs up) ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
hanging up a leg will automatically hang up b leg, if it doesnt you still have unresolved nat problems. make your phone register every 20 sec to keep the sip path alive. On Thu, Mar 25, 2010 at 1:20 PM, Madovsky <infos@madovsky.org> wrote:
I use an audio stream relayed by a client that has no audio DSP card. so when I stop to talk, no UDP audio packet is sent to FS. Do you know a trick to send fake UDP packet to keep the call alive ? (I need to keep rtp-timeout short to hang up leg B when leg A hangs up)
Thanks
F
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
ok. no nat is used. but for sure it could be a client problem ----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Thursday, March 25, 2010 2:29 PM Subject: Re: [Freeswitch-users] send fake UDP packets hanging up a leg will automatically hang up b leg, if it doesnt you still have unresolved nat problems. make your phone register every 20 sec to keep the sip path alive. On Thu, Mar 25, 2010 at 1:20 PM, Madovsky <infos@madovsky.org> wrote: I use an audio stream relayed by a client that has no audio DSP card. so when I stop to talk, no UDP audio packet is sent to FS. Do you know a trick to send fake UDP packet to keep the call alive ? (I need to keep rtp-timeout short to hang up leg B when leg A hangs up) Thanks F _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Anthony Minessale
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Brian West
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Madovsky