As Iñaki said ....
and one more - the is no difference if the routing in the SIP proxy is
done stateless or stateful.
regards
klaus
Stagg Shelton wrote:
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@63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-b4657f0*
To: <sip:+16783832765@8.17.32.184:5060>;tag=as40da5b97*
Call-ID: ATLMGC0720080823144655027771(a)209.244.63.15
<mailto:ATLMGC0720080823144655027771@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@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@8.17.32.184 SIP/2.0*
From: Anonymous
<sip:restricted@63.209.207.135>;tag=88cfd13f-13c4-48b022df-dcaa3e6a-b4657f0*
To: <sip:+16783832765@8.17.32.184:5060>;tag=as40da5b97*
Call-ID: ATLMGC0720080823144655027771(a)209.244.63.15
<mailto:ATLMGC0720080823144655027771@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@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
>>
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