[Kamailio-Users] Having Problem routing an ACK
Stagg Shelton
stagg at sheltonjohns.com
Mon Aug 25 16:41:37 CEST 2008
This is exactly what I told the carrier engineer about the problem.
The contact in the 200 OK is my Asterisk PBX the RURI in the ACK
(returning from the carrier) is my OpenSER. I informed the carrier
that according to my interpretation of the RFC 3261 the RURI in the
ACK MUST match the contact in the 200 OK with SDP.
The carrier interop engineer sent me the following remark after I
notified them of the issue, and began pressing them on the way their
ACK was created.
== BEGIN ==
The Request URI field is used for routing of initial dialog messages.
If state is not kept in [Open]SER by using the tm module, the SIP
message can be routed based on the VIA or contact header, or worst
case follow the same routing rules based on the original Request URIs,
just like the original INVITE message did.
== END ==
After some subsequent communication where I again questioned the way
in which their system creates the ACK. I received the below from them.
== BEGIN ==
All I was saying is that based on the way your system handles messages
we send to it, your statement about changing the Request-URIs of ACKs
based on values from the SDP of the 200 OK is not the case. I believe
that some systems do however update the ACK’s Request-URI based on the
Contact header field from the 200 response, but most system’s don’t
route the ACK based on its Request-URI when keeping state.
I have another question based on our use of SER internally. When we
send the initial INVITE, the Request-URI is set as the number at (@)
the address of your [Open]SER proxy. SER will then re-write the
request URI of the INVITE based on the logic in the local ser.cfg file
and any local location/alias database entries used in that logic and
forward the message to the destination, in this case the PBX, while
adding a Record-Route header with LR set to ensure SER stays in the
dialog in terms of signaling as well as adding a VIA header with a
branch tag used to mark the dialog for stateful processing (when the
TM module is used). In the case of stateful processing, these values
are used to maintain the same signaling path for subsequent messages
within the same dialog.
When stateless routing is used via the SL module, all new requests
(including an ACK to a 200 response) should follow the same routing
logic as the initial request. Therefore, when the ACK arrives at your
SER proxy, the same routing logic should be applied to the ACK as was
the original INVITE, and it should be forwarded on to the correct
destination. If this is not the case, then either there must be some
kind of state being tracked or the logic has been written to handle
ACKs differently on purpose as this would be the only way that the
handling and routing (especially the computation of the final request-
uri) would be different for an ACK from an INVITE.
== END ==
At this point I am ready to address any issue with my OpenSER
configuration that can be identified, but if the problem is actually
due to the ACK not being constructed correctly, I have to take off
the technical hat, and put on the business hat, and try to get them to
look at their systems, and push their vendors for support in this issue.
Thank You
Stagg Shelton
On Aug 25, 2008, at 9:52 AM, Klaus Darilion wrote:
> Hi!
>
> AFAIS the client is buggy (or is there a NAT ALG/Firewall between
> client and SIP proxy?). Compare the Contact header in the 200 OK and
> the request URI in the ACK. They MUST be the same!!!
>
> regards
> klaus
>
>
> U +0.000315 8.17.32.184:5060 -> 63.209.207.135:5060
> SIP/2.0 200 OK*
> Via: SIP/2.0/UDP 63.209.207.135:5060;branch=z9hG4bK-8921-48b022df-
> dcaa3e6a-2f5ec169*
> Record-Route: <sip:8.17.32.184;lr=on;did=952.4d684275>*
> From: Anonymous
> <sip:restricted at 63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-
> b4657f0*
> To: <sip:+16783832765 at 8.17.32.184:5060>;tag=as40da5b97*
> Call-ID: ATLMGC0720080823144655027771 at 209.244.63.15*
> CSeq: 1 INVITE*
> User-Agent: Asterisk PBX*
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY*
> Supported: replaces*
> Contact: <sip:+16783832765 at 8.17.32.19>*
> Content-Type: application/sdp*
> Content-Length: 180*
>
>
>
> U +0.072541 63.209.207.135:5060 -> 8.17.32.184:5060
> ACK sip:+16783832765 at 8.17.32.184 SIP/2.0*
> From: Anonymous
> <sip:restricted at 63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-
> b4657f0*
> To: <sip:+16783832765 at 8.17.32.184:5060>;tag=as40da5b97*
> Call-ID: ATLMGC0720080823144655027771 at 209.244.63.15*
> CSeq: 1 ACK*
> Via: SIP/2.0/UDP 63.209.207.135:5060;branch=z9hG4bK-8922-48b022e5-
> dcaa5757-3884948f*
> Max-Forwards: 15*
> Contact: <sip:restricted at 63.209.207.135:5060;transport=udp>*
> Route: <sip:8.17.32.184;lr;did=952.4d684275>*
> Content-Length: 0*
>
>
>
>
> Stagg Shelton schrieb:
>> Thanks again Iñaki. I am attaching siptrace.txt file. I can see
>> that there appears to be something odd with the ACKs in that they
>> appear to be sent from my openser back to my openser in a loop
>> until the max forwards is reached.
>> ------------------------------------------------------------------------
>> Thank you for your help.
>> Stagg Shelton.
>> On Aug 23, 2008, at 10:08 AM, Iñaki Baz Castillo wrote:
>>> El Sábado, 23 de Agosto de 2008, Stagg Shelton escribió:
>>>> Iñaki,
>>>>
>>>> Thank you for your response. I have enabled the siptrace module in
>>>> openser. The data in the mysql table only shows the trace
>>>> between the
>>>> carrier and openser. Can I submit a pcap file that shows all of
>>>> the
>>>> SIP communication that occured during the call.
>>>
>>> Hi, you don't need to enable siptrace. Just install "ngrep" and do:
>>>
>>> ngrep -d any -P '*' -W byline -T port 5060
>>>
>>>
>>> --
>>> Iñaki Baz Castillo
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> ------------------------------------------------------------------------
>> _______________________________________________
>> Users mailing list
>> Users at lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
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