Dear list, I am currently trying to send faxes via T.38 using PJSIP (newest version 2.3) with Asterisk 13.0.2. After having configured PJSIP, I have seen several things the cause of which I would like to know.
1) Ports and IP addresses which PJSIP bind to I have configured one transport like that: [tr_wZCMk5MvC2ATNzAr] type = transport protocol = udp bind = 192.168.20.48 Nevertheless, PJSIP binds to more ports and IP addresses than expected: root@spock:~# netstat -apnv | grep asterisk udp 0 0 192.168.20.48:5060 0.0.0.0:* 21416/asterisk udp 0 0 0.0.0.0:42415 0.0.0.0:* 21416/asterisk udp 0 0 0.0.0.0:48565 0.0.0.0:* 21416/asterisk [SNIP] This is on a box which has one physical NIC which has configured multiple IP addresses by ethernet aliasing: eth0 Link encap:Ethernet HWaddr 02:01:01:01:05:01 inet addr:192.168.20.238 Bcast:192.168.20.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:32321283 errors:0 dropped:0 overruns:0 frame:0 TX packets:171282095 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX bytes:9690993944 (9.0 GiB) TX bytes:244294378305 (227.5 GiB) eth0:1 Link encap:Ethernet HWaddr 02:01:01:01:05:01 inet addr:192.168.20.48 Bcast:192.168.20.255 Mask:255.255.255.0 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 [and so on, 10 IP addresses] So what is the meaning of the additional ports PJSIP is opening, and why does it open these on all IP addresses? By the way, I already have tried to make sure that it's really PJSIP which opens these. After all, I can tell for sure that they are NOT opened if I use chan_sip instead of PJSIP with an otherwise identical software version (I am compiling myself so I was able to produce two flavors of Asterisk which are identical except that one uses chan_sip, the other one chan_pjsip). I furthermore have compiled an additional Asterisk / PJSIP flavor with as few modules, channels etc. as possible, but this flavor still opens the additional ports. 2) Wrong owner in SDP (o= line) I think this problem relates to the first one. I am currently unable to send a fax, and I suspect this is due to the fact that Asterisk / PJSIP produces a wrong owner record. A typical INVITE: No. Time Source Destination Protocol Length Info 9225 7.503015 192.168.20.48 xx.xxx.xx.xxx SIP/SDP 886 Request: INVITE sip:004982349663...@fpbx.de | Frame 9225: 886 bytes on wire (7088 bits), 886 bytes captured (7088 bits) Ethernet II, Src: MS-NLB-PhysServer-01_01:01:05:01 (02:01:01:01:05:01), Dst: D-Link_03:a4:18 (00:1b:11:03:a4:18) Internet Protocol Version 4, Src: 192.168.20.48 (192.168.20.48), Dst: xx.xxx.xx.xxx (xx.xxx.xx.xxx) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol (INVITE) Request-Line: INVITE sip:00498234xxxx...@itsp.de SIP/2.0 Method: INVITE Request-URI: sip:00498234xxxx...@itsp.de Request-URI User Part: 0049823xxxxxxx Request-URI Host Part: itsp.de [Resent Packet: False] Message Header Via: SIP/2.0/UDP 79.211.71.113:5060;rport;branch=z9hG4bKPj7afca7e1-0c3b-494f-978a-844fa19cfc4a Transport: UDP Sent-by Address: yy.yyy.yy.yyy Sent-by port: 5060 RPort: rport Branch: z9hG4bKPj7afca7e1-0c3b-494f-978a-844fa19cfc4a From: <sip:77748zb...@fpbx.de>;tag=4e855dd1-4a8c-41a9-9524-038d32c08ce3 SIP from address: sip:usern...@itsp.de SIP from address User Part: username SIP from address Host Part: itsp.de SIP from tag: 4e855dd1-4a8c-41a9-9524-038d32c08ce3 To: <sip:004982349663...@fpbx.de> SIP to address: sip:00498234xxxx...@itsp.de SIP to address User Part: 00498234xxxxxxx SIP to address Host Part: itsp.de Contact: <sip:usern...@yy.yyy.yy.yyy> Contact URI: sip:usern...@yy.yyy.yy.yyy Contact URI User Part: username Contact URI Host Part: yy.yyy.yy.yyy Call-ID: ad46d131-91ab-44bd-8b7e-40551b7fd8e5 CSeq: 20417 INVITE Sequence Number: 20417 Method: INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 238 Message Body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 928891384 928891384 IN IP4 192.168.20.238 Owner Username: - Session ID: 928891384 Session Version: 928891384 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.20.238 Session Name (s): Asterisk Connection Information (c): IN IP4 yy.yyy.yy.yyy Connection Network Type: IN Connection Address Type: IP4 Connection Address: yy.yyy.yy.yyy Time Description, active time (t): 0 0 Session Start Time: 0 Session Stop Time: 0 Media Description, name and address (m): audio 11544 RTP/AVP 0 101 Media Type: audio Media Port: 11544 Media Protocol: RTP/AVP Media Format: ITU-T G.711 PCMU Media Format: DynamicRTP-Type-101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Format: 0 MIME Type: PCMU Sample Rate: 8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Format: 101 MIME Type: telephone-event Sample Rate: 8000 Media Attribute (a): fmtp:101 0-16 Media Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): maxptime:150 Media Attribute Fieldname: maxptime Media Attribute Value: 150 Media Attribute (a): sendrecv Note that in the SDP part it claims the Owner/Creator (o=) to be 192.168.20.238 which is the main IP address of the box (eth0), but not the one where Asterisk / PJSIP should bind to. So, I've got two questions here: First, how do I tell Asterisk / PJSIP which IP address it should use as the owner (o=) IP address (I didn't see anything in the docs which would allow for that), and secondly, could a wrong owner be the reason for the ITSP to hang up immediately after the T.38 re-invite? In other words, does a wrong owner harm at all? Thank you very much for any ideas, Recursive -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users