Randall Garner Posted May 7, 2012 Report Share Posted May 7, 2012 We receive about 2-4 of these RTP Timout Errors a month. Below is the most recent error. Can anyone shine some light into what is causing this error for me? Thanks in advance! 2012/5/6 09:30:33 Tx: udp:24.96.139.170:5060 (388 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 24.96.139.170:5060;branch=z9hG4bK4FA6.8B19.8D414442.AA8B6018.00001BCD Via: SIP/2.0/UDP 10.170.172.21:5060;branch=z9hG4bK4FA6.8B19.8D414442 From: WIRELESS CALLER <sip:6189263170@10.170.172.21:5060>;tag=8B19.8D41 To: <sip:8502345271@10.170.172.170:7117>;tag=4b066a3184 Call-ID: 0026.9E9B.B4F2.4FA6.8B19.0D42 CSeq: 4442 INVITE Content-Length: 0 2012/5/6 09:30:33 Tx: udp:24.96.139.170:5060 (937 bytes) SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 24.96.139.170:5060;branch=z9hG4bK4FA6.8B19.8D414442.AA8B6018.00001BCD Via: SIP/2.0/UDP 10.170.172.21:5060;branch=z9hG4bK4FA6.8B19.8D414442 From: WIRELESS CALLER <sip:6189263170@10.170.172.21:5060>;tag=8B19.8D41 To: <sip:8502345271@10.170.172.170:7117>;tag=4b066a3184 Call-ID: 0026.9E9B.B4F2.4FA6.8B19.0D42 CSeq: 4442 INVITE Contact: <sip:8502345271@10.36.1.18:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 237 v=0 o=- 56015 56015 IN IP4 10.36.1.18 s=- c=IN IP4 10.36.1.18 t=0 0 m=audio 53962 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/5/6 09:30:33 Rx: udp:24.96.139.170:5060 (586 bytes) PRACK sip:8502345271@10.36.1.18:5060 SIP/2.0 From: WIRELESS CALLER <sip:6189263170@10.170.172.21:5060>;tag=8B19.8D41 To: <sip:8502345271@10.170.172.170:7117>;tag=4b066a3184 Call-ID: 0026.9E9B.B4F2.4FA6.8B19.0D42 CSeq: 4443 PRACK Max-Forwards: 70 Via: SIP/2.0/UDP 24.96.139.170:5060;branch=z9hG4bK4FA6.8B19.8D414443.AA8B6018.00001BCD Via: SIP/2.0/UDP 10.170.172.21:5060;branch=z9hG4bK4FA6.8B19.8D414443 Contact: <sip:6189263170@24.96.139.170:5060> RAck: 1 4442 INVITE Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE Content-Length: 0 2012/5/6 09:30:33 Tx: udp:24.96.139.170:5060 (478 bytes) SIP/2.0 200 Ok Via: SIP/2.0/UDP 24.96.139.170:5060;branch=z9hG4bK4FA6.8B19.8D414443.AA8B6018.00001BCD Via: SIP/2.0/UDP 10.170.172.21:5060;branch=z9hG4bK4FA6.8B19.8D414443 From: WIRELESS CALLER <sip:6189263170@10.170.172.21:5060>;tag=8B19.8D41 To: <sip:8502345271@10.170.172.170:7117>;tag=4b066a3184 Call-ID: 0026.9E9B.B4F2.4FA6.8B19.0D42 CSeq: 4443 PRACK Contact: <sip:8502345271@10.36.1.18:5060;transport=udp> User-Agent: snom-PBX/2011-4.2.1.4025 Content-Length: 0 2012/5/6 09:30:53 Tx: udp:24.96.139.170:5060 (897 bytes) SIP/2.0 200 Ok Via: SIP/2.0/UDP 24.96.139.170:5060;branch=z9hG4bK4FA6.8B19.8D414442.AA8B6018.00001BCD Via: SIP/2.0/UDP 10.170.172.21:5060;branch=z9hG4bK4FA6.8B19.8D414442 From: WIRELESS CALLER <sip:6189263170@10.170.172.21:5060>;tag=8B19.8D41 To: <sip:8502345271@10.170.172.170:7117>;tag=4b066a3184 Call-ID: 0026.9E9B.B4F2.4FA6.8B19.0D42 CSeq: 4442 INVITE Contact: <sip:8502345271@10.36.1.18:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.1.4025 Content-Type: application/sdp Content-Length: 237 v=0 o=- 56015 56015 IN IP4 10.36.1.18 s=- c=IN IP4 10.36.1.18 t=0 0 m=audio 53962 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv 2012/5/6 09:30:53 Rx: udp:24.96.139.170:5060 (427 bytes) ACK sip:8502345271@10.36.1.18:5060 SIP/2.0 From: WIRELESS CALLER <sip:6189263170@10.170.172.21:5060>;tag=8B19.8D41 To: <sip:8502345271@10.170.172.170:7117>;tag=4b066a3184 Call-ID: 0026.9E9B.B4F2.4FA6.8B19.0D42 CSeq: 4442 ACK Max-Forwards: 70 Via: SIP/2.0/UDP 24.96.139.170:5060;branch=z9hG4bK4FA6.8B19.8D414444.AA8B6018.00001BCD Via: SIP/2.0/UDP 10.170.172.21:5060;branch=z9hG4bK4FA6.8B19.8D414444 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted May 8, 2012 Report Share Posted May 8, 2012 Well, if the PBX does not receive any media, it assumes that the connection went down. In plain words, thats "one way audio". Unfortunately in VoIP that is not uncommon, especially when terminating calls over the Internet, where you can not always be sure where the call is being terminated. The user should have the one-way audio experience as well, maybe you can check next time with the involved user if he really experienced one-way audio. Quote Link to comment Share on other sites More sharing options...
Randall Garner Posted May 9, 2012 Author Report Share Posted May 9, 2012 I will follow your recommendation and ask the users ASAP next time I receive this timeout. Thanks for your help! Quote Link to comment Share on other sites More sharing options...
Jeremy Isherwood Posted July 24, 2012 Report Share Posted July 24, 2012 I have the same fault; I have remote users, who did connect to the pbx via a vpn, but since moving to an external wan address i have no audio on either in or out going calls. Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 24, 2012 Report Share Posted July 24, 2012 Assuming the calls were fine when running on the VPN, the new setup might have some networking issues. The firewalls on your network or the remote network could be blocking the UDP(RTP) traffic. Quote Link to comment Share on other sites More sharing options...
Jeremy Isherwood Posted July 25, 2012 Report Share Posted July 25, 2012 Yes after reading hundreds of posts, I guessed that was the case. My question now is, so what ports need to be open? Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 25, 2012 Report Share Posted July 25, 2012 Please take a look at this wiki page http://wiki.snomone.com/index.php?title=RTP_Ports. Quote Link to comment Share on other sites More sharing options...
Jeremy Isherwood Posted July 25, 2012 Report Share Posted July 25, 2012 Thanks but the page has no content. it just says: There is currently no text in this page. You can search for this page title in other pages, or search the related logs. do you have any other information? Quote Link to comment Share on other sites More sharing options...
Jeremy Isherwood Posted July 25, 2012 Report Share Posted July 25, 2012 Oh hold on, it said that because your link had a . I can see the page now, but it does not give me any details of the actual port numbers I need to open on my routers... Quote Link to comment Share on other sites More sharing options...
katerina Posted July 25, 2012 Report Share Posted July 25, 2012 Try this one too: http://wiki.snomone.com/index.php?title=Ports_to_open Quote Link to comment Share on other sites More sharing options...
pbx support Posted July 25, 2012 Report Share Posted July 25, 2012 Did you really navigate to "Admin > Settings > Ports" and looked at the "Port Range Start" and "Port Range end" as the wiki link says? Quote Link to comment Share on other sites More sharing options...
Jeremy Isherwood Posted July 25, 2012 Report Share Posted July 25, 2012 Yes I have, but just needed confirmation that it was only the port numbers listed there that were at fault. As it turns out, my supplier here in the UK (Spitfire) own software was not allowing the open ports I requested, in fact their system would automatically close the ports even through they opened them. I had been working all day today with their support. So tomorrow they are giving me the login details to open the ports and I will see if this changes my no audio problem. Thanks for your understanding in my numerous posts. Quote Link to comment Share on other sites More sharing options...
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