[SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

[Alex Villací­s Lasso]

El 01/09/14 05:15, Daniel-Constantin Mierla escribió:
>
> On 29/08/14 23:58, Andres wrote:
>> On 8/29/14, 1:42 PM, Alex Villací­s Lasso wrote:
>>> Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170:
>>>
>>> IP 198.58.101.75.5060 > 201.234.196.170.5060
>>> INVITE sip:*43 at 201.234.196.170:5060 SIP/2.0
>>> Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport
>>> Max-Forwards: 70
>>> From: "9002" <sip:9002 at 198.58.101.75>;tag=as0bc522a9
>>> To: <sip:*43 at 201.234.196.170:5060>
>>> Contact: <sip:9002 at 198.58.101.75:5060>
>>> Call-ID: 2c14c21f5052a74a78ca4ab736657b00 at 198.58.101.75:5060
>>> CSeq: 102 INVITE
>>> User-Agent: FPBX-2.8.1(1.8.20.0)
>>> Date: Fri, 29 Aug 2014 18:23:17 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
>>> Supported: replaces, timer
>>> Content-Type: application/sdp
>>> Content-Length: 299
>>>
>>> v=0
>>> o=root 521741684 521741684 IN IP4 198.58.101.75
>>> s=Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3
>>> c=IN IP4 198.58.101.75
>>> t=0 0
>>> m=audio 16426 RTP/AVP 0 8 3 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> IP 201.234.196.170.5060 > 198.58.101.75.5060
>>> SIP/2.0 100 trying -- your call is important to us
>>> Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
>>> From: "9002" <sip:9002 at 198.58.101.75>;tag=as0bc522a9
>>> To: <sip:*43 at 201.234.196.170:5060>
>>> Call-ID: 2c14c21f5052a74a78ca4ab736657b00 at 198.58.101.75:5060
>>> CSeq: 102 INVITE
>>> Server: kamailio (4.1.5 (x86_64/linux))
>>> Content-Length: 0
>>>
>>>
>>> IP 201.234.196.170.5060 > 198.58.101.75.5060
>>> SIP/2.0 200 OK
>>> Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7a792c1e;rport=5060
>>> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
>>> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as0bc522a9;vsf=SRoZSkpbSEZbLF1YW0dGeB8ICB8bDxsxMDEuNzU-;nat=yes>
>>> From: "9002" <sip:9002 at 198.58.101.75>;tag=as0bc522a9
>>> To: <sip:*43 at 201.234.196.170:5060>;tag=as2798a3b9
>>> Call-ID: 2c14c21f5052a74a78ca4ab736657b00 at 198.58.101.75:5060
>>> CSeq: 102 INVITE
>>> Server: Asterisk PBX 11.12.0
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
>>> Supported: replaces, timer
>>> Session-Expires: 1800;refresher=uas
>>> Contact: <sip:*43 at 127.0.0.1:5080;alias=127.0.0.1~5080~1>
>>> Content-Type: application/sdp
>>> Require: timer
>>> Content-Length: 305
>>>
>>> v=0
>>> o=root 159029581 159029581 IN IP4 201.234.196.170
>>> s=Asterisk PBX 11.12.0
>>> c=IN IP4 201.234.196.170
>>> t=0 0
>>> m=audio 18446 RTP/AVP 0 8 3 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:3 GSM/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>> a=sendrecv
>>> a=nortpproxy:yes
>>>
>>>
>>> According to a strict interpretation of the SIP RFC, which address should the machine at 198.58.101.75 use to send the subsequent ACK? Which field(s) are to be used to extract said address? I am trying to understand an issue of a missing ACK between 
>>> 201.234.196.17x and a different public IP, with the only difference that the other IP is not running Asterisk. For the exchange shown above, 201.234.196.170 receives an ACK, but I want to know whether the packets correctly indicate the address for the 
>>> ACK, or whether the Asterisk at 198.58.101.75 is compensating for a malformed packet.
>>>
>> Regardless of what the first hop of the ACK is going to be, the Contact Field in the SIP 200 OK is telling 198.58.101.75 that the ACK should be directed to 127.0.0.1 which is probably not what you want.
>>
> In this case, the contact in 200ok is the last hop of the ACK, because the 200ok includes Record-Route headers. Therefore the caller has to send the ACK to last Record-Route address in 200ok.
>
> That is also private/non-routable address in internet and I expect is not what it is desired, considering the other endpoints work with public IP.
>
> I guess the sip server is running on a natted system (e.g., amazon-cloud-like). You may want to add 'advertise' address to listen core parameter in order to use public ip in signaling packets -- see core cookbook for more on listen parameter.
>
> Cheers,
> Daniel
>

The test system is running inside a local network and has a local address of 192.168.2.18.

So, it is true that the remote asterisk is covering up for a mistake in my kamailio config?