Summary:	ASTERISK-26011: [patch]PJSIP: add "via_addr", "via_port", "call_id" to contacts
Reporter:	Alexei Gradinari (alexei gradinari)	Labels:
Date Opened:	2016-05-10 15:59:43	Date Closed:	2016-07-05 11:42:32
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Resources/res_pjsip Resources/res_pjsip_registrar
Versions:	13.9.0	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contacts. Added new fields ViaAddress,CallID to AMI event ContactStatus. [edit by Rusty - adding Alexei's further explanation] To manually manage/provisioning remote (behind the NAT) SIP devices it's necessary to know internal ip-address. We could get internal ip-address using old-SIP AMI command 'SIPShowPeer' For example Reg-Contact : sip:user@192.168.1.2:5060 Using PJSIP we couldn't because res_pjsip_nat rewrites contact's address before AMI event is sent. To find out internal ip-address 2 SIP headers can help: Via and Call-Id.
Comments:	By: Asterisk Team (asteriskteam) 2016-05-10 15:59:44.448-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Walter Doekes (wdoekes) 2016-05-11 03:18:49.803-0500 Hi. Thanks for the patches. If you could you add the URL to the gerrit review in the "Reviewboard URL" field in the future, it helps. Thanks :) I did not get a chance too look at your patch in gerrit at this point, because it is down (daily reload), but I wonder what the purpose of this patch is. res_pjsip_nat rewrites the contact: ok, and then. When do we want the via_addr/via_port/callid, why do we want it? What would we do with it? Is it worth storing all of that for every registered peer always? By: Alexei Gradinari (alexei gradinari) 2016-05-11 15:46:07.359-0500 Hi Walter, If you click on "Gerrit Reviews" tab it has all gerrit reviews for this topic. To manually manage/provisioning remote (behind the NAT) SIP devices it's necessary to know internal ip-address. We could get internal ip-address using old-SIP AMI command 'SIPShowPeer' For example Reg-Contact : sip:user@192.168.1.2:5060 Using PJSIP we couldn't because res_pjsip_nat rewrites contact's address before AMI event is sent. To find out internal ip-address 2 SIP headers can help: Via and Call-Id.