Re: [asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short I would very much like to get a wireshark trace (pcap file) of that session to try to understand this message. I don't think the RTCP read to short affects your communication - that is propably another issue. But since I've become a bit occupied with RTCP lately, I would like to see what causes this message. If you have the oppurtunity, or someone else that sees this message in your Aterisk, please send me the packet trace off list, directly to my personal e-mail o...@edvina.net. Thanks for the assistance! /O -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
, ts 423545829, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042259, ts 007920, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042260, ts 008160, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010682, ts 423546069, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042261, ts 008400, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010683, ts 423546309, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042262, ts 008640, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010684, ts 423546549, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042263, ts 008880, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010685, ts 423546789, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042264, ts 009120, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010686, ts 423547029, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042265, ts 009360, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010687, ts 423547269, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042266, ts 009600, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010688, ts 423547509, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042267, ts 009840, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010689, ts 423547749, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010690, ts 423547989, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042268, ts 010080, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042269, ts 010560, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010691, ts 423548229, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042270, ts 010800, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010692, ts 423548469, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010693, ts 423548709, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042271, ts 011040, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010694, ts 423548949, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042272, ts 011280, len 24) Got RTP packet from 192.168.1.64:16392 (type 04, seq 010695, ts 423549189, len 24) Sent RTP packet to 192.168.1.64:16392 (type 04, seq 042273, ts 011520, len 24) --- On Fri, 1/29/10, Alexandru Oniciuc alexandru.onic...@trivenet.it wrote: From: Alexandru Oniciuc alexandru.onic...@trivenet.it Subject: [asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, January 29, 2010, 7:15 AM Hello Wassim, server side you can check the RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the file isn’t there, here are the defaults: ;[general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 You can even debug the RTP : CLI rtp debug ip xxx.xxx.xxx.xxx(linksys) Asterisk listens on one of those ports(rtp.conf ones) when a call is initiated. The same does your Linksys GW: it will listen only on the RTP configured ports. Check the firewall between the VoIP server and the Linsys GW and check the firewall on the Asterisk server. Debugging SIP you can see which ports are involved. There might be other problems, maybe because you are trying to directly pass the call from one peer(let’s say an external voice provider) to the other(linksys). In that case careinvite=no is be your friend. Regards, Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me please. -Inline Attachment Follows- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
The ring isn't fake :] The Linksys GW isn't dissing, is just responding to an INVITE. The problem is that you have problem passing voice. In other words: check RTP ports settings on server client or the firewall rules. Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 17:38 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: I have a linksys voip gateway connecting to an asterisk server ,when i dial a call from the linksys gateway to asterisk , i see repeated messages of a RTP errors ,and at same time i hear fake ring on the linksys , This is wht i see on asterisk console : -- Executing [9613070...@direct:1] Set(SIP/03070741-088bd470, CALLERID(number)=96170707070) in new stack -- Executing [9613070...@direct:2] Dial(SIP/03070741-088bd470, SIP/usa/9613070741) in new stack -- Called usa/9613070741 [Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 -- SIP/usa-08906450 is ringing -- Call on SIP/usa-08906450 left from hold -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 [Jan 28 18:17:50] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:53] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:57] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: rtp.c:883 ast_rtcp_read: RTCP Read too short
Hello Wassim, server side you can check the RTP ports configured in rtp.conf which you will find in /etc/asterisk/. If the file isn't there, here are the defaults: ;[general] ; ; RTP start and RTP end configure start and end addresses ; ; Defaults are rtpstart=5000 and rtpend=31000 You can even debug the RTP : CLI rtp debug ip xxx.xxx.xxx.xxx(linksys) Asterisk listens on one of those ports(rtp.conf ones) when a call is initiated. The same does your Linksys GW: it will listen only on the RTP configured ports. Check the firewall between the VoIP server and the Linsys GW and check the firewall on the Asterisk server. Debugging SIP you can see which ports are involved. There might be other problems, maybe because you are trying to directly pass the call from one peer(let's say an external voice provider) to the other(linksys). In that case careinvite=no is be your friend. Regards, Alex Da: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Per conto di wassim darwich Inviato: giovedì 28 gennaio 2010 21:41 A: asterisk-users@lists.digium.com Oggetto: [asterisk-users] rtp.c:883 ast_rtcp_read: RTCP Read too short Hi: Firewall is disabled ,so no need to worry about firewall,but i dont know where to check rtp settings and what do i need to search for ,can you guide me please. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
