[OpenSER-Users] OpenSER with TLS - method BYE = protocol/port mismatch ??
Daniel-Constantin Mierla
daniel at voice-system.ro
Tue Aug 21 14:21:05 CEST 2007
Hello,
please send the request uri as well and the details about the IP
communication (source/destination address, port).
use:
ngrep -qt -W byline port 5060
the replace the IPs with something suggestive to protect your privacy.
Daniel
On 08/21/07 14:28, David Loh wrote:
> Hi Daniel,
>
> Thanks for the reply, here's my INVITE and BYE packet, I tcpdump-ed
> and and use wireshark to export as txt, so the format might be abit off.
>
> 1) Invite from OpenSER to Asterisk via port 5060, asterisk return a
> 200. And the call successfully established
> ------------------------------------------------------------------------
> Record-Route: <sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>
> Record-Route:
> <sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
> Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bKe42e.ff11ae32.0;i=52
> Via: SIP/2.0/TLS
> %UAC_LAN_IP%:23323;received=%UAC_WAN_IP%;branch=z9hG4bK-d87543-2820ee009262b500-1--d87543-;rport=3135
> Max-Forwards: 69
> Contact: <sip:davidloh@%UAC_WAN_IP%:3135;transport=TLS>
> To: "001"<sip:001@%SIP_Domain%>
> From: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
> Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: eyeBeam release 1004p stamp 31962
> Content-Length: 239
> ------------------------------------------------------------------------
>
> 2) Here's the BYE, from Asterisk to OpenSER.
> ------------------------------------------------------------------------
> Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport
> Route:
> <sip:%OpenSER_IP%;r2=on;lr;ftag=1247871f;nat=yes>,<sip:%OpenSER_IP%:5061;transport=tls;r2=on;lr;ftag=1247871f;nat=yes>
> From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
> To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
> Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
> CSeq: 102 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> ------------------------------------------------------------------------
>
> 3) And then, OpenSER passing the BYE to UAC...
> ------------------------------------------------------------------------
> Record-Route: <sip:%OpenSER_IP%;lr;ftag=as0492a9d0>
> Via: SIP/2.0/UDP %OpenSER_IP%;branch=z9hG4bK5e.1195fd8.0
> Via: SIP/2.0/UDP %Asterisk_IP%:5060;branch=z9hG4bK0ca1ab09;rport=5060
> From: "001"<sip:001@%SIP_Domain%>;tag=as0492a9d0
> To: "ser"<sip:davidloh@%SIP_Domain%>;tag=1247871f
> Call-ID: Zjc4OWJhZGNiMDUxNjQ5Y2I2ZDU3NTc4NGY2Y2RmOGQ.
> CSeq: 102 BYE
> User-Agent: Asterisk PBX
> Max-Forwards: 69
> Content-Length: 0
> ------------------------------------------------------------------------
>
> I think the "protocol/port mismatch" happened right after (3) because
> from my ethereal trace,
> after short while (perhaps timeout), Asterisk will re-initiate a "BYE"
> and pass to OpenSER (packet #2), and OpenSER will forward it to UAC
> (packet #3),
> but somehow UAC doesn't response an "ACK" to OpenSER that why from my
> ethereal trace, packet #2 and #3 kept looping.
>
> Any thoughts ?
>
>
> Thanks in advance.
>
> Regards,
> David Loh
>
>
>
> Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> can you paste the BYE here (you can catch it with ngrep if it is not
>> coming via tls, e.g., from asterisk)? Also, the invite going to
>> asterisk will be good.
>>
>> Cheers,
>> Daniel
>>
>>
>> On 08/20/07 07:50, David Loh wrote:
>>> Hi All,
>>>
>>> Good day, I'm new to this mailing list :-)
>>>
>>> I've setup an OpenSER server with TLS implementation, so far I
>>> tested everything works fine,
>>> but there's strange problem for "BYE" request, from the openser's
>>> log it always complaint:
>>> "Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
>>> protocol/port mismatch"
>>>
>>> Here's my implementation:
>>> UA -> OpenSER -> Asterisk (as RTP) -> OpenSER -> UA
>>> On OpenSER I've two port opened which is Udp 5060 and TCP 5061 (TLS).
>>> My UA is Eyebeam v1.5 and the CA certificate (self-signed) already
>>> installed into IE "Trusted RootCA Certificates".
>>> So far I've tested p2p and pstn calls (and call to playback
>>> balance/asterisk), but the problem is unless UA disconnect the call
>>> (initiate BYE),
>>> otherwise if remote party (p2p/pstn/asterisk) initiate BYE request,
>>> my UA will forever stay connected until I disconnect the call manually.
>>>
>>> So far I've identified the BYE request from remote party will go
>>> into Loose Route, and I've place a xlog to display the R-Uri..
>>> here's the log from openser: (notes: if UA dial '001' will playback
>>> his/her balance from asterisk)
>>> -------------------------------------------------------------------------------------------------------------
>>>
>>> Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE from client
>>> sip:%user%@%domain% (%UA_IP%) - p2p call
>>> Aug 19 22:58:31 ser /sbin/openser[16189]: DBG: INVITE
>>> F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
>>> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
>>> sip:%user%@%domain% (%UA_IP%)
>>> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE from client
>>> sip:%user%@%domain% (%UA_IP%) - p2p call
>>> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: INVITE
>>> F-[sip:%user%@%domain%] T-[sip:001@%domain%] IP-[%UA_IP%] SUCCESSFUL
>>> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG:
>>> [sip:%user%@%domain%](%UA_IP%) ACK [sip:001@%domain%] relayed LOOSE
>>> ROUTE
>>> Aug 19 22:58:32 ser /sbin/openser[16189]: DBG: ruri
>>> [sip:001@%Asterisk_IP%] relayed LOOSE ROUTE
>>> Aug 19 22:58:40 ser /sbin/openser[16171]: DBG:
>>> [sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
>>> LOOSE ROUTE
>>> Aug 19 22:58:40 ser /sbin/openser[16171]: DBG: ruri
>>> [sip:%user%@%domain%:%UA_Port%] relayed LOOSE ROUTE
>>> Aug 19 22:58:40 ser /sbin/openser[16171]: WARNING:get_send_socket:
>>> protocol/port mismatch <<<=============
>>> Aug 19 22:58:41 ser /sbin/openser[16177]: DBG:
>>> [sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
>>> LOOSE ROUTE
>>> Aug 19 22:58:41 ser /sbin/openser[16177]: DBG: ruri
>>> [sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
>>> Aug 19 22:58:42 ser /sbin/openser[16181]: DBG:
>>> [sip:001@%domain%](%Asterisk_IP%) BYE [sip:%user%@%domain%] relayed
>>> LOOSE ROUTE
>>> Aug 19 22:58:42 ser /sbin/openser[16181]: DBG: ruri
>>> [sip:%user%@%UA_IP%:%UA_Port%] relayed LOOSE ROUTE
>>> ....... (repeat the last 4 lines N times )
>>> -------------------------------------------------------------------------------------------------------------
>>>
>>> Below is my openser.cfg, loose route section
>>> -------------------------------------------------------------------------------------------------------------
>>>
>>> ###############
>>> # Loose route #
>>> ###############
>>> if(loose_route()) {
>>> if(has_totag() && (is_method("INVITE") || is_method("ACK"))) {
>>> if(nat_uac_test("19") || search("^Route:.*;nat=yes")) {
>>> fix_nated_contact();
>>> }
>>> }
>>>
>>> xlog("L_INFO", "DBG: [$fu]($si) $rm [$tu] relayed LOOSE ROUTE");
>>> xlog("L_INFO", "DBG: ruri [$ru] relayed LOOSE ROUTE");
>>> t_relay();
>>>
>>> exit;
>>> }
>>> -------------------------------------------------------------------------------------------------------------
>>>
>>>
>>> Had anyone encountered the same problems before ? Do you mind to
>>> share your experience with me ?
>>> Your help are greatly appreciated, thanks in advance.
>>>
>>> Regards,
>>> David Loh
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at openser.org
>>> http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>
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