[asterisk-bugs] [JIRA] (ASTERISK-29194) PJSIP
Joshua C. Colp (JIRA)
noreply at issues.asterisk.org
Thu Dec 3 15:38:16 CST 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29194?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Joshua C. Colp updated ASTERISK-29194:
--------------------------------------
Assignee: Mark Murawski
Status: Waiting for Feedback (was: Triage)
The "rewrite_contact" option controls SIP signaling. It changes the SIP contact to be the actual source IP address and port. It has no influence over RTP.
The "rtp_symmetric" option would be the only one that does, which will send RTP to the source IP address and port from which it is received. Until RTP is received then Asterisk will send media to the IP address and port provided in the SDP.
As well the "local_net" option appears to be working fine. The IP address placed into the SDP and signaling from Asterisk is what I would expect.
> PJSIP
> -----
>
> 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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