[asterisk-bugs] [JIRA] (ASTERISK-29194) PJSIP NAT - rtp_symmetric not working
Mark Murawski (JIRA)
noreply at issues.asterisk.org
Thu Dec 3 18:44:16 CST 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29194?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=252938#comment-252938 ]
Mark Murawski commented on ASTERISK-29194:
------------------------------------------
Thanks Josh!
Not a bug
Resolution: Remove per-endpoint transport=
> PJSIP NAT - rtp_symmetric not working
> -------------------------------------
>
> Key: ASTERISK-29194
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29194
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_pjsip
> Affects Versions: 16.15.0
> Reporter: Mark Murawski
> Assignee: Mark Murawski
>
> In certain situations PJSIP does not respect local_net settings
> Given the following setup:
> {noformat}
> 21005 | endpoint/force_rport | yes
> 21005 | endpoint/rewrite_contact | yes
> 21005 | endpoint/rtp_symmetric | yes
> [transport-udp-lo]
> type = transport
> protocol = udp
> bind = 127.0.0.1:5060
> external_media_address = 127.0.0.1
> external_signaling_address = 127.0.0.1
> external_signaling_port = 5060
> allow_reload = yes
> tos = cs3
> cos = 3
> [transport-udp-tun0]
> type = transport
> protocol = udp
> bind = 10.1.2.20:5060
> external_media_address = 10.1.2.20
> external_signaling_address = 10.1.2.20
> external_signaling_port = 5060
> allow_reload = yes
> tos = cs3
> cos = 3
> local_net = 10.1.2.0/24
> [transport-udp-tun1]
> type = transport
> protocol = udp
> bind = 10.3.2.20:5060
> external_media_address = 10.3.2.20
> external_signaling_address = 10.3.2.20
> external_signaling_port = 5060
> allow_reload = yes
> tos = cs3
> cos = 3
> local_net = 10.3.2.0/24
> {noformat}
> And traffic coming into transport-udp-tun1
> {noformat}
> 15:36:45.656238 IP 10.3.2.1.5060 > 10.3.2.20.5060: SIP: INVITE sip:*1234 at 10.3.2.20:5060;user=phone SIP/2.0
> E`..B%..>...
> ...
> ...........INVITE sip:*1234 at 10.3.2.20:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.50.206;branch=z9hG4bKe661cfb7BD8F7712
> From: "21005" <sip:21005 at 10.3.2.20>;tag=9BA63BB1-914B7CEC
> To: <sip:*1234 at 10.3.2.20;user=phone>
> CSeq: 1 INVITE
> Call-ID: cfd9c5fd-64b01f78-fa86f683 at 192.168.50.206
> Contact: <sip:21005 at 192.168.50.206>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.15.1009
> Accept-Language: en
> Supported: 100rel,replaces
> Allow-Events: conference,talk,hold
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 298
> v=0
> o=- 1607027804 1607027804 IN IP4 192.168.50.206
> s=Polycom IP Phone
> c=IN IP4 192.168.50.206
> t=0 0
> a=sendrecv
> m=audio 2228 RTP/AVP 9 0 8 18 127
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 15:36:45.657358 IP 10.3.2.20.5060 > 10.3.2.1.5060: SIP: SIP/2.0 401 Unauthorized
> E`.Ry. at .@...
> ...
> ........>"#SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 192.168.50.206;rport=5060;received=10.3.2.1;branch=z9hG4bKe661cfb7BD8F7712
> Call-ID: cfd9c5fd-64b01f78-fa86f683 at 192.168.50.206
> From: "21005" <sip:21005 at 10.3.2.20>;tag=9BA63BB1-914B7CEC
> To: <sip:*1234 at 10.3.2.20;user=phone>;tag=z9hG4bKe661cfb7BD8F7712
> CSeq: 1 INVITE
> WWW-Authenticate: Digest realm="asterisk",nonce="1607027805/57d9f055573ea588c2beb1f03c8e2bca",opaque="345e83785480fbf0",algorithm=md5,qop="auth"
> Server: Asterisk PBX 16.15.0
> Content-Length: 0
> 15:36:45.698544 IP 10.3.2.1.5060 > 10.3.2.20.5060: SIP: ACK sip:*1234 at 10.3.2.20:5060;user=phone SIP/2.0
> E`.GB&..>. .
> ...
> ........3j.ACK sip:*1234 at 10.3.2.20:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.50.206;branch=z9hG4bKe661cfb7BD8F7712
> From: "21005" <sip:21005 at 10.3.2.20>;tag=9BA63BB1-914B7CEC
> To: <sip:*1234 at 10.3.2.20;user=phone>;tag=z9hG4bKe661cfb7BD8F7712
> CSeq: 1 ACK
> Call-ID: cfd9c5fd-64b01f78-fa86f683 at 192.168.50.206
> Contact: <sip:21005 at 192.168.50.206>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.15.1009
> Accept-Language: en
> Max-Forwards: 70
> Content-Length: 0
> 15:36:45.698568 IP 10.3.2.1.5060 > 10.3.2.20.5060: SIP: INVITE sip:*1234 at 10.3.2.20:5060;user=phone SIP/2.0
> E`..B'..>..n
> ...
> ..........6INVITE sip:*1234 at 10.3.2.20:5060;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.50.206;branch=z9hG4bK3302ce1eA5B33949
> From: "21005" <sip:21005 at 10.3.2.20>;tag=9BA63BB1-914B7CEC
> To: <sip:*1234 at 10.3.2.20;user=phone>
> CSeq: 2 INVITE
> Call-ID: cfd9c5fd-64b01f78-fa86f683 at 192.168.50.206
> Contact: <sip:21005 at 192.168.50.206>
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
> User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.15.1009
> Accept-Language: en
> Supported: 100rel,replaces
> Allow-Events: conference,talk,hold
> Authorization: Digest username="21005", realm="asterisk", nonce="1607027805/57d9f055573ea588c2beb1f03c8e2bca", qop=auth, cnonce="hTWM3tpAkuCZjNC", nc=00000001, opaque="345e83785480fbf0", uri="sip:*1234 at 10.3.2.20:5060;user=phone", response="7e4d360452dbfdf02c7297786403e104", algorithm=MD5
> Max-Forwards: 70
> Content-Type: application/sdp
> Content-Length: 298
> v=0
> o=- 1607027804 1607027804 IN IP4 192.168.50.206
> s=Polycom IP Phone
> c=IN IP4 192.168.50.206
> t=0 0
> a=sendrecv
> m=audio 2228 RTP/AVP 9 0 8 18 127
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:127 telephone-event/8000
> 15:36:45.701474 IP 10.3.2.20.5060 > 10.3.2.1.5060: SIP: SIP/2.0 100 Trying
> E`..y3 at .@...
> ...
> ...........SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.50.206;rport=5060;received=10.3.2.1;branch=z9hG4bK3302ce1eA5B33949
> Call-ID: cfd9c5fd-64b01f78-fa86f683 at 192.168.50.206
> From: "21005" <sip:21005 at 10.3.2.20>;tag=9BA63BB1-914B7CEC
> To: <sip:*1234 at 10.3.2.20;user=phone>
> CSeq: 2 INVITE
> Server: Asterisk PBX 16.15.0
> Content-Length: 0
> 15:36:45.716260 IP 10.3.2.20.5060 > 10.3.2.1.5060: SIP: SIP/2.0 200 OK
> E`..y4 at .@...
> ...
> .........v.SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.50.206;rport=5060;received=10.3.2.1;branch=z9hG4bK3302ce1eA5B33949
> Call-ID: cfd9c5fd-64b01f78-fa86f683 at 192.168.50.206
> From: "21005" <sip:21005 at 10.3.2.20>;tag=9BA63BB1-914B7CEC
> To: <sip:*1234 at 10.3.2.20;user=phone>;tag=9e82c96e-4b35-4ac9-a055-21970e96c5fd
> CSeq: 2 INVITE
> Server: Asterisk PBX 16.15.0
> Contact: <sip:1.2.3.4:5060>
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Content-Type: application/sdp
> Content-Length: 255
> v=0
> o=- 1607027804 1607027806 IN IP4 10.3.2.20
> s=Asterisk
> c=IN IP4 10.3.2.20
> t=0 0
> m=audio 16430 RTP/AVP 0 8 127
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:127 telephone-event/8000
> a=fmtp:127 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> {noformat}
> ------------------
> The following occurs:
> {noformat}
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030925, ts 006080, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030926, ts 006240, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030927, ts 006400, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030928, ts 006560, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030929, ts 006720, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030930, ts 006880, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030931, ts 007040, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030932, ts 007200, len 000160)
> Sent RTP packet to 192.168.50.206:2228 (type 00, seq 030933, ts 007360, len 000160)
> {noformat}
> Instead of sending to the expected rewrite_contact of 10.3.2.1
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