Defect #2401
closed
SIP retransmissions - packet timeout
0%
Description
Hi
I have this error since I set up IPtables for the first time.
The call comes in, is being answered and then is hanged up by the server.
[2014-02-02 20:37:27] WARNING6900: chan_sip.c:4174 retrans_pkt: Retransmission timeout reached on transmission 73b4bd613dec2e597d5dfb9159a95646@176.9.39.206 for seqno 102 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6401ms with no response
[2014-02-02 20:37:27] WARNING6900: chan_sip.c:4203 retrans_pkt: Hanging up call 73b4bd613dec2e597d5dfb9159a95646@176.9.39.206 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
Within IPtables I opened the following IP addresses
81.23.228.140 ACCEPT udp -- 81.23.228.140 0.0.0.0/0 udp dpts:5060:5069
81.23.228.129 ACCEPT udp -- 81.23.228.129 0.0.0.0/0 multiport dports 5060,5061,5062,5063,5064,5065,5066,5067,5068,5069
81.23.228.150 ACCEPT udp -- 81.23.228.150 0.0.0.0/0 multiport dports 5060,5061,5062,5063,5064,5065,5066,5067,5068,5069
85.17.186.7 ACCEPT udp -- 85.17.186.7 0.0.0.0/0 multiport dports 5060,5061,5062,5063,5064,5065,5066,5067,5068,5069
I am not very experienced at reading the SIP trace, but it seems an ACK is missing.
SIP session 6d98b38743d0bb735cdba9cf24de6e48@176.9.39.206 node 11 20:41:05
Should I add 176.9.39.206 to allow in my IPtables?
What else can be / did I do wrong?
thanks
Updated by R Mont almost 11 years ago
I should perhaps add that I have no problems with other SIP accounts that are registered, only sip2sip.
Updated by Tijmen de Mes almost 11 years ago
We see the ACK being sent out on the 20:41 call (packet 7 on node07). After it left our platform we don't know what happens with it. You should check if it arrives and if it does why it is ignored.
Updated by R Mont almost 11 years ago
After a good few hours of analysing this i found the fix that made it work...would like your opinion though.
ISSUE:
Incoming call to a number at anveo.com (sip.de.anveo.com) is forwarded to URi number@sip2sip.info
My asterisk has a trunk registered to sip2sip.info
Call arrives to my system, phones start ringing, I answer and there is two way voice for 6 seconds after which the call is dropped.
1. The ACK does not arrive at my asterisk system (behind NAT), even if I take down its iptables.
2. When I register a trunk directly to anveo (note: UDP 5010), the ACK arrives and the call continues and is not dropped.
3. Trunks from other providers are also working fine.
1&2&3 means probably that my router is OK and does not block ACK
So I changed externip in my asterisk system to 192.168.x.x instead of to the WAN address.
This solved the issue. My question is, why ?
Here are the two ACK's from the working and the not working situation:
The working situation shows an extra Route line, what is this?
Working
ACK sip:s@my.ip.my.ip:1024 SIP/2.0
Via: SIP/2.0/UDP 85.17.186.7:5060;branch=z9hG4bK94d.6859e993.3
Via: SIP/2.0/UDP 176.9.39.206:9119;received=176.9.39.206;branch=z9hG4bK0aa6cdde;rport=9119
Route: <sip:81.23.228.129;lr;ftag=as5247b571;did=97e.70070862>
From: "31123456789" <sip:31123456789@176.9.39.206:9119>;tag=as5247b571
To: <sip:31000000000@sip2sip.info>;tag=as1133e9e2
Contact: <sip:31123456789@176.9.39.206:9119>
Call-ID: 345ec4380562c3a156be3828254ca9e0@176.9.39.206
CSeq: 102 ACK
User-Agent: Anveo Server v10.3
Max-Forwards: 69
Content-Length: 0
Not working
ACK sip:s@my.ip.my.ip:5060 SIP/2.0
Via: SIP/2.0/UDP 85.17.186.7:5060;branch=z9hG4bK99a6.2dc5ff83.2
Via: SIP/2.0/UDP 176.9.39.206:9119;received=176.9.39.206;branch=z9hG4bK33cd6a65;rport=9119
From: "0041229600770" <sip:0041229600770@176.9.39.206:9119>;tag=as14dff3c3
To: <sip:97293720000@sip2sip.info>;tag=as06a14621
Contact: <sip:0041229600770@176.9.39.206:9119>
Call-ID: 6d98b38743d0bb735cdba9cf24de6e48@176.9.39.206
CSeq: 102 ACK
User-Agent: Anveo Server v10.3
Max-Forwards: 69
Content-Length: 0