[asterisk-bugs] [JIRA] (ASTERISK-29194) PJSIP
Kevin Harwell (JIRA)
noreply at issues.asterisk.org
Thu Dec 3 15:34:16 CST 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29194?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Kevin Harwell updated ASTERISK-29194:
-------------------------------------
Description:
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
was:
In certain situations PJSIP does not respect local_net settings
Given the following setup:
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
And traffic coming into transport-udp-tun1
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
------------------
The following occurs:
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)
Instead of sending to the expected rewrite_contact of 10.3.2.1
> 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
>
> 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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