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.
- Invite from OpenSER to Asterisk via port 5060, asterisk return a
- 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
- 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
- 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
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