| Summary: | ASTERISK-27079: PJSIP puts invalid data in SDP when using external_media_address | ||||
| Reporter: | snuffy (snuffy) | Labels: | |||
| Date Opened: | 2017-06-22 05:30:40 | Date Closed: | 2020-06-30 15:57:30 | ||
| Priority: | Major | Regression? | No | ||
| Status: | Closed/Complete | Components: | Resources/res_pjsip | ||
| Versions: | GIT | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Debian 8.1, ESXi VM, 2 network interfaces Asterisk GIT-master-db5e269 | Attachments: | ( 0) answer33.pcap ( 1) answer-debug22.pcap ( 2) answer-fail-5.pcap ( 3) answer-new11.pcap ( 4) ast-full-2min.log ( 5) my-test.log ( 6) pjsip-trimmed.conf | ||
| Description: | I am getting an issue when SIP timers are turned on, could be related to use of external_signal_address
session_expires=1800 (30min) According to my PCAP output: * INVITE created for outbound call (session starts with timer set at 1800) * Correct SDP IP information is set * 15 minutes in, new INVITE is send out * Correct SDP IP information is set * Session timer expires (30 minutes), new INVITE is sent out * Garbage is set for SDP IP * Other side replies with 488 - Invalid SDP information * Call is terminated 15 minutes later by Asterisk The invalid SDP: {noformat} INVITE sip:0410551660@202.10.4.137:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.237.181.230:5060;rport;branch=z9hG4bKPjdea05bde-0e5a-4bae-bc3d-79ca37a53680 From: <sip:0383708000@10.21.16.253>;tag=62136bf6-1bf5-494c-9a30-19cdc11e91b0 To: <sip:0410551660@202.10.4.137>;tag=97615 Contact: <sip:asterisk@10.237.181.230:5060> Call-ID: e22eb82b-b822-4c84-bb1b-8d045e9848c2 CSeq: 15537 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 120;refresher=uac Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX GIT-master-db5e269 Content-Type: application/sdp Content-Length: 171 v=0 o=- 1615031026 1615031026 IN IP4 {noformat} | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2017-06-22 05:30:41.300-0500 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: snuffy (snuffy) 2017-06-22 05:42:36.691-0500 Versions Tested: Master from 2 days ago. Date: Tue Jun 20 18:18:19 2017 -0500 By: Sean Bright (seanbright) 2017-06-26 10:24:58.774-0500 Can you attach your PJSIP configuration please? Redact as needed. By: snuffy (snuffy) 2017-06-26 20:26:12.795-0500 Now attached pjsip.conf Internal person is from the 3230 number on IP of 10.21.16.103 Asterisk is on the 10.21.16.253 address /24 subnet. Exists routes for forwarding traffic out to the SIP provider listed, default GW is for reaching other local site ip addresses which are not listed in this config. 0.0.0.0 10.21.16.254 0.0.0.0 UG 0 0 0 eth0 10.21.16.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0 10.236.2.0 10.237.181.229 255.255.255.0 UG 0 0 0 eth1 10.237.181.228 0.0.0.0 255.255.255.252 U 0 0 0 eth1 202.10.4.0 10.237.181.229 255.255.255.0 UG 0 0 0 eth1 202.92.72.0 10.237.181.229 255.255.255.0 UG 0 0 0 eth1 210.87.44.6 10.237.181.229 255.255.255.255 UGH 0 0 0 eth1 By: Rusty Newton (rnewton) 2017-06-28 08:54:37.229-0500 Can you include a debug log that has pjsip logging enabled as well? It is a little easier to correlate debug with the SIP flow when they are together. Having the pcap is nice too, so thanks for that. By: snuffy (snuffy) 2017-06-28 22:23:08.866-0500 Debug log from pjsip? I thought my-test.log had pjsip debug info in it? Or are you looking for something else ? By: Rusty Newton (rnewton) 2017-06-29 09:45:54.093-0500 I was looking for the PJSIP logger output, that is, the SIP traffic. You've got the pcap which shows that side of it, but it is nice to have the SIP traffic in the debug log to help identify the chronology and relationship of debug messages to specific points in the call flow. By: snuffy (snuffy) 2017-06-29 11:10:38.953-0500 new upload By: snuffy (snuffy) 2017-06-29 11:11:14.576-0500 new upload 'pjsip set logger host xxxx' By: Rusty Newton (rnewton) 2017-07-01 13:20:58.928-0500 Perfect, thanks. By: Joshua C. Colp (jcolp) 2020-06-30 15:57:30.625-0500 Closing out in favor of ASTERISK-28973. | ||||