Am 01.07.2010 22:33, schrieb Ole Kaas:
Hello,
The guy operating the Asterisk at the far end have been inspecting the traces I've sent him. Hen notes that the ACK relayed doesn't entirely comply to the standard. The values of "branch" in the VIA header of the relayed ACK response incorrectly have the value "0" (also rport in the VIA header added by their Asterisk is modified). He suggest setting the syn_branch parameter to get the "branch" right. I've found this thread about it:
This some question pops up again and again. The RFC is not 100% clear in this aspect. It states that any RFC3261 compatible UA/proxy must add a unique branch parameter, starting with the magic cookie. (Thus, Kamailio should use a unique branch in ACK).
Further RFC3261 states, that if a user agent receives a SIP request where the topmost Via header contains a branch parameter which does not start with the magic cookie, the UA has to accept the request and uses RFC2543-based transaction matching.
Thus, Kamailio - when used as RFC3261 complient proxy - should use a unique branch parameter. On the other hand, RFC3261 compatible clients should accept branch=0 too.
I just wonder why this Asterisk-based Gateway does not accept branch=0. I have several Asterisk versions in use and never experienced any issue with branch=0.
regards Klaus
I have put syn_branch=0 in the config and "branch" now have a valid value. I don't quite buy it - that the incorrect value of "branch" should be the culprit. But I'll activate tcpdump again and wait to see if the dropped calls has vanished.
/Ole
Den 01/07/2010 kl. 09.12 skrev Klaus Darilion:
Hi!
Kamailio behaves correct in this trace and I couldn't spot an abvious error in the trace.
Nevertheless there are 2 problems:
- Asterisk Gateway does not receive/accept the ACK 2. Asterisk PBX
does not retransmit ACK
reg. 1: ask the gateway operator if he sees the ACK in the Asterisk log, and if yes the error message of Asterisk why this ACK is not accepted. If the ACK is not seen in the Asterisk console then maybe it gets dropped by a buggy firewall.
reg 2: login to your Asterisk PBX and verify if Asterisk receives the 200 OK, and maybe spot some log message why it does not trigger ACK retransmissions.
regards Klaus
Am 30.06.2010 23:33, schrieb Ole Kaas:
Den 30/06/2010 kl. 01.23 skrev Iñaki Baz Castillo:
2010/6/29 Ole Kaasobk@tet.dk:
Hi Klaus,
I've mailed pcap dump to you directly for further inspection.
Hi, it's much better if you capture a trace with "ngrep -Wbyline -t -q port 5060" and paste it in a new mail by replacing your public IP's with other values. Then all the people here could help you rather than asking for private help to a specific member of the maillist.
You are right. But maybe it was something (obvious) that could be resolved quickly and I could post an update here on the list. The original log was inadequate - Klaus was a great help, with suggestions to obtain new log. So here it is attached and anonymized with all ip addresses (and stuff) converted to private adresses. The Kamailio server is multi homed and the two networks are non-routable (I use rtpproxy to bridge media). Our Asterisk PBX is version 1.4.26.1 and the Asterisk Gateway is 1.6.1 (or 1.6.0 - cant remember and not under my control). Kamailio is version 3.0.0.
Looking at the trace, it seems the problem starts with the ACK not being received by the Asterisk Gateway which then resends the OK. The OK is relayed back to the originating Asterisk PBX which seems to ignore the retransmission. In fact it seems that Kamailio is routing and relaying the sip packets correctly. However, it seems that the problem only exists between Asterisk and Kamailio. I have other pbx'es (3CX) connecting to Kamailio and I have no evidence that the problem happens with those. I have another trace where the call comes from one of the Asterisk Gateways and is routed back to one of the other Asterisk Gateways. The result is the same - the OK's are ignored by Asterisk.
/Ole