[Kamailio-Users] Need Assistance Tracking Down A SIP Signaling Problem
Daniel-Constantin Mierla
miconda at gmail.com
Thu Sep 24 23:25:54 CEST 2009
There is something wrong upstream, in the carrier network.
One observation -- the ACK is looping on .189.92 -- there are two Via
headers.
Since the contact from 200ok is lost, the proxy does not how to handle
the ACK which is part of INVITE dialog. You have to ask the carrier to
fix it in its side. It is not much you can do unless you try some hacks,
e.g., use a htable to store target asterisk based on callid, detect bad
ACK and forward them using the IP from htable.
However, you probably have to do the same hacks for BYE (and other
in-dialog requests), when coming from carrier.
Cheers,
Daniel
On 24.09.2009 23:11 Uhr, Geoffrey Mina wrote:
> Thanks. Here is the relevant bits of the failed SIP session
>
>
> -------------------------------------------------
> INVITE Carrier To Kamailio:
> -------------------------------------------------
> INVITE sip:+18889160750 at 208.72.190.171:5060 SIP/2.0
> Record-Route: <sip:208.72.189.92;lr=on;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.190.201;lr=on>
> t: <sip:+18889160750 at 208.72.190.201:5060;user=phone>
> f: <sip:+19496776128 at 10.10.100.124:5060;user=phone>;tag=f3ec973c-251ccd56-7c640a0a
> Remote-Party-Id:
> <sip:+19496776128 at 10.10.100.124:5060;user=phone>;screen=yes;id-type=subscriber;party=calling;privacy=off
> i: 01263185-ac-251ccd56 at 10.10.100.124
> CSeq: 1027265 INVITE
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.0
> Via: SIP/2.0/UDP 208.72.190.201;branch=z9hG4bK4ffc.0f42a522.0
> v: SIP/2.0/UDP 10.10.100.124:5060;branch=z9hG4bK009cc5864d6d3965
> Max-Forwards: 68
> m: <sip:+19496776128 at 10.10.100.124:5060;user=phone>
> k: replaces
> c: application/sdp
> Accept: application/sdp
> Accept-Encoding:
> Accept-Language: en
> User-Agent: Lucent-Universal-Gateway
> l: 253
>
> -------------------------------------------------
> INVITE Kamailio To Asterisk :
> -------------------------------------------------
> INVITE sip:+18889160750 at 208.72.190.183:5060 SIP/2.0
> Record-Route: <sip:208.72.190.171;lr;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.189.92;lr=on;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.190.201;lr=on>
> t: <sip:+18889160750 at 208.72.190.201:5060;user=phone>
> f: <sip:+19496776128 at 10.10.100.124:5060;user=phone>;tag=f3ec973c-251ccd56-7c640a0a
> Remote-Party-Id:
> <sip:+19496776128 at 10.10.100.124:5060;user=phone>;screen=yes;id-type=subscriber;party=calling;privacy=off
> i: 01263185-ac-251ccd56 at 10.10.100.124
> CSeq: 1027265 INVITE
> Via: SIP/2.0/UDP 208.72.190.171;branch=z9hG4bK4ffc.659965f3.0
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.0
> Via: SIP/2.0/UDP 208.72.190.201;branch=z9hG4bK4ffc.0f42a522.0
> v: SIP/2.0/UDP 10.10.100.124:5060;branch=z9hG4bK009cc5864d6d3965
> Max-Forwards: 67
> m: <sip:+19496776128 at 10.10.100.124:5060;user=phone>
> k: replaces
> c: application/sdp
> Accept: application/sdp
> Accept-Encoding:
> Accept-Language: en
> User-Agent: Lucent-Universal-Gateway
> l: 253
>
>
> -------------------------------------------------
> OK Asterisk To Kamailio:
> -------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 208.72.190.171;branch=z9hG4bK4ffc.659965f3.0;received=208.72.190.171
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.0
> Via: SIP/2.0/UDP 208.72.190.201;branch=z9hG4bK4ffc.0f42a522.0
> Via: SIP/2.0/UDP 10.10.100.124:5060;branch=z9hG4bK009cc5864d6d3965
> Record-Route: <sip:208.72.190.171;lr;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.189.92;lr=on;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.190.201;lr=on>
> From: <sip:+19496776128 at 10.10.100.124:5060;user=phone>;tag=f3ec973c-251ccd56-7c640a0a
> To: <sip:+18889160750 at 208.72.190.201:5060;user=phone>;tag=as233f7874
> Call-ID: 01263185-ac-251ccd56 at 10.10.100.124
> CSeq: 1027265 INVITE
> User-Agent: G-Tel v1.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:+18889160750 at 208.72.190.183>
> Content-Type: application/sdp
> Content-Length: 244
>
> -------------------------------------------------
> OK Kamilio to Asterisk:
> -------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.0
> Via: SIP/2.0/UDP 208.72.190.201;branch=z9hG4bK4ffc.0f42a522.0
> Via: SIP/2.0/UDP 10.10.100.124:5060;branch=z9hG4bK009cc5864d6d3965
> Record-Route: <sip:208.72.190.171;lr;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.189.92;lr=on;ftag=f3ec973c-251ccd56-7c640a0a>
> Record-Route: <sip:208.72.190.201;lr=on>
> From: <sip:+19496776128 at 10.10.100.124:5060;user=phone>;tag=f3ec973c-251ccd56-7c640a0a
> To: <sip:+18889160750 at 208.72.190.201:5060;user=phone>;tag=as233f7874
> Call-ID: 01263185-ac-251ccd56 at 10.10.100.124
> CSeq: 1027265 INVITE
> User-Agent: G-Tel v1.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:+18889160750 at 208.72.190.183>
> Content-Type: application/sdp
> Content-Length: 244
>
>
> -------------------------------------------------
> ACK Carrier to Kamailio:
> -------------------------------------------------
> ACK sip:208.72.190.171;lr;ftag=f3ec973c-251ccd56-7c640a0a SIP/2.0
> t: <sip:+18889160750 at 208.72.190.201:5060;user=phone>;tag=as233f7874
> f: <sip:+19496776128 at 10.10.100.124:5060;user=phone>;tag=f3ec973c-251ccd56-7c640a0a
> i: 01263185-ac-251ccd56 at 10.10.100.124
> CSeq: 1027265 ACK
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.2
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.2
> Via: SIP/2.0/UDP 208.72.190.201;branch=z9hG4bK4ffc.0f42a522.2
> v: SIP/2.0/UDP 10.10.100.124:5060;branch=z9hG4bK009cc59b4f4d6d16
> Max-Forwards: 67
> User-Agent: Lucent-Universal-Gateway
> l: 0
>
> -------------------------------------------------
> ACK Kamilio to Kamailio: (we are no into the loop)
> -------------------------------------------------
> ACK sip:208.72.190.171;lr;ftag=f3ec973c-251ccd56-7c640a0a SIP/2.0
> t: <sip:+18889160750 at 208.72.190.201:5060;user=phone>;tag=as233f7874
> f: <sip:+19496776128 at 10.10.100.124:5060;user=phone>;tag=f3ec973c-251ccd56-7c640a0a
> i: 01263185-ac-251ccd56 at 10.10.100.124
> CSeq: 1027265 ACK
> Via: SIP/2.0/UDP 208.72.190.171;branch=z9hG4bK4ffc.659965f3.2
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.2
> Via: SIP/2.0/UDP 208.72.189.92;branch=z9hG4bK4ffc.321fc1c6.2
> Via: SIP/2.0/UDP 208.72.190.201;branch=z9hG4bK4ffc.0f42a522.2
> v: SIP/2.0/UDP 10.10.100.124:5060;branch=z9hG4bK009cc59b4f4d6d16
> Max-Forwards: 66
> User-Agent: Lucent-Universal-Gateway
> l: 0
>
>
> On Thu, Sep 24, 2009 at 4:51 PM, Daniel-Constantin Mierla
> <miconda at gmail.com> wrote:
>
>> Hello,
>>
>> On 24.09.2009 22:36 Uhr, Geoffrey Mina wrote:
>>
>>> Hello,
>>> I am wondering if anyone can help me determine what the problem is
>>> with some SIP signaling. I have two environments and in both
>>> scenarios my configuration and topology are almost identical...
>>> although I am dealing with two different carriers upstream.
>>>
>>> In my environments, I have Kamailio (1.5) sitting in front of a
>>> multitude of Asterisk machines. I am using the dispatcher module to
>>> distribute INVITE requests across the network. I am doing some
>>> interop with a new carrier and we are struggling a bit with some
>>> looping scenarios. They are sending invites to my Kamailio server, I
>>> am forwarding to asterisk.
>>>
>>> On the one that is not working, I am seeing the following in sip_trace
>>>
>>> INVITE (from carrier)
>>> INVITE (to asterisk)
>>> 100 TRYING (from asterisk)
>>> 200 OK (from asterisk)
>>> 200 OK (to carrier)
>>> ACK (from carrier) - this is where the loop starts. Kamailio sends
>>> the ACK to itself until the "max-hops" is reached... then it dies
>>> ACK (from itself to itself)
>>> ACK (from itself to itself)
>>> ACK (from itself to itself)
>>> ...
>>> 200 OK (from asterisk - because it never got ACK)
>>> 200 OK (to carrier)
>>> ACK (from carrier) - again the loop.
>>> ACK (from itself to itself)
>>>
>>>
>>> The only thing I can see that is different between the two carriers
>>> (working and non-working) is that the working carrier appears to send
>>> the ACK with an RURI equivalent to the Contact: header from the 200
>>> OK.
>>>
>> this is the correct behavior for loose routing.
>>
>> it looks like the one of the devices (very likely the carrier) is doing bad
>> record routing handling. There might be a strict router combined with a
>> loose router.
>>
>> I could really follow the sip messages in the excel you sent, maybe you can
>> paste text here in email the invite, the 200ok and ACK captured on kamailio
>> server. Then is easier to troubleshoot.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla
>> * Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
>> * http://www.asipto.com/index.php/sip-router-masterclass/
>>
>>
>>
--
Daniel-Constantin Mierla
* Kamailio SIP Masterclass, Nov 9-13, 2009, Berlin
* http://www.asipto.com/index.php/sip-router-masterclass/
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