Hello!  I am searching for the reason why PhonerLite is changing the connection IP in the SDP on a re-invite after 401 Unauthorized challenge.  The first invite has the correct gateway connection IP (172.16.55.20) in the invite but after the 401 challenge response the re-invite has the private IP (the wrong IP 192.168.0.113) in the SDP.   

I've checked my server config using another softphone that does not do this so I can only assume there is a setting I am missing that I need to change.   Thank you for any help. 
2018/01/07 22:27:04.711631 172.16.55.20:5060 -> 10.170.58.33:5060
INVITE sip:1000@10.170.58.33 SIP/2.0
Via: SIP/2.0/UDP 172.16.55.20:5060;branch=z9hG4bK807c847091f2e711baadfb07bf501c7d;rport
From: "8088" <sip:8088@10.170.58.33>;tag=2377651663
To: <sip:1000@10.170.58.33>
Call-ID: 807C8470-91F2-E711-BAAC-FB07BF501C7D@192.168.0.113
CSeq: 5 INVITE
Contact: <sip:8088@172.16.55.20:5060;gr=009D6C60-91F2-E711-BAA5-FB07BF501C7D>
Content-Type: application/sdp
Mime-Version: 1.0
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, gruu, timer
P-Preferred-Identity: <sip:8088@10.170.58.33>
Content-Length:   547

v=0
o=- 2752989180 1 IN IP4 172.16.55.20
s=SIPPER for PhonerLite
c=IN IP4 172.16.55.20
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:11 L16/44100
a=rtpmap:118 L16/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:2569897716
a=sendrecv





2018/01/07 22:27:04.712145 10.170.58.33:5060 -> 172.16.55.20:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1baadfb07bf501c7d
Call-ID: 807C8470-91F2-E711-BAAC-FB07BF501C7D@192.168.0.113
From: "8088" <sip:8088@10.170.58.33>;tag=2377651663
To: <sip:1000@10.170.58.33>;tag=z9hG4bK807c847091f2e711baadfb07bf501c7d
CSeq: 5 INVITE
WWW-Authenticate: Digest  6b8295e043a06",algorithm=md5,qop="auth"
Server: Rogue v00.00.00
Content-Length:  0




2018/01/07 22:27:04.736878 172.16.55.20:5060 -> 10.170.58.33:5060
ACK sip:1000@10.170.58.33 SIP/2.0
Via: SIP/2.0/UDP 172.16.55.20:5060;branch=z9hG4bK807c847091f2e711baadfb07bf501c7d;rport
From: "8088" <sip:8088@10.170.58.33>;tag=2377651663
To: <sip:1000@10.170.58.33>;tag=z9hG4bK807c847091f2e711baadfb07bf501c7d
Call-ID: 807C8470-91F2-E711-BAAC-FB07BF501C7D@192.168.0.113
CSeq: 5 ACK
Content-Length: 0



2018/01/07 22:27:04.737630 172.16.55.20:5060 -> 10.170.58.33:5060
INVITE sip:1000@10.170.58.33 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.113:5060;branch=z9hG4bK808aab7791f2e711baaefb07bf501c7d;rport
From: "8088" <sip:8088@10.170.58.33>;tag=2377651663
To: <sip:1000@10.170.58.33>
Call-ID: 807C8470-91F2-E711-BAAC-FB07BF501C7D@192.168.0.113
CSeq: 6 INVITE
Contact: <sip:8088@192.168.0.113:5060;gr=009D6C60-91F2-E711-BAA5-FB07BF501C7D>
Authorization: Digest username="8088", realm="asterisk", nonce="1515382024/705f880ceead38aa14d58960e0fc6baa", uri="sip:1000@10.170.58.33", response="fadef30082e36f4eefb0a393b035c1d6", algorithm=MD5, cnonce="808aab7791f2e711baadfb07bf
1c7d", opaque="1306b8295e043a06", qop=auth, nc=00000001
Content-Type: application/sdp
Mime-Version: 1.0
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, gruu, timer
P-Preferred-Identity: <sip:8088@10.170.58.33>
Content-Length:   549

v=0
o=- 2752989180 1 IN IP4 192.168.0.113
s=SIPPER for PhonerLite
c=IN IP4 192.168.0.113
t=0 0
m=audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:11 L16/44100
a=rtpmap:118 L16/

Aha!  I did not know of the debug window.  That is a great tool.  Phonerlite is a fantastic soft phone.  The debug showed that the SDP was not changing from the softphone but led me to looking at my router.  There was a SIP ALG enabled that was changing the SDP on the initial invite.  Between disabling that and enabling the UPnP NAT on Phonerlite my problem has been solved and I now have good two way audio.  Many thanks for the tip of the debug window!