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RTP Timout


Randall Garner
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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

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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.

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  • 2 months later...

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.

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