[asterisk-bugs] [JIRA] (ASTERISK-29938) PJSIP Rewritng headers along with protocol

Vijo Jose (JIRA) noreply at issues.asterisk.org
Fri Feb 25 08:00:07 CST 2022 

Vijo Jose created ASTERISK-29938:
------------------------------------

             Summary: PJSIP Rewritng headers along with protocol
                 Key: ASTERISK-29938
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29938
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip
    Affects Versions: 16.20.0
         Environment: Asterisk 16.20.0
Centos 7.9

            Reporter: Vijo Jose
            Severity: Blocker


Am using SIP with TLS for my SIP User registration Also I have a SIP Proxy configured for user registration request forwarding towards my asterisk box.

Users who are registering in the Local network.
SIP Proxy have public Internet
And Asterisk Box also have Public Internet
>From the USer machine, we have the connectivity to Proxy and From proxy to Asterisk as well. And my sip user will register with a softphone using UDP to Asterisk domain with the proxy address and my proxy send to asterisk with TLS. Here I have everything work perfectly with Chan_sip

Now I need to change that need chan_pjsip and try for registration - But the registration working and the calls are failing. But in SIP it was working fine.

I have a user config for a sip. conf

[123123]
username=123123
type=friend
secret=Asterisk at 123123 note that this is NOT a secure password
host=dynamic
qualify=yes
context=from-pstn
dtmfmode=rfc2833
disallow=all
allow=alaw
transport=tls
encryption=yes

And the same user converts to pjsip and below is pjsip.conf
[123123]
username=123123

[123123]
type=aor
max_contacts=1
maximum_expiration=3600
default_expiration=120

[123123]
type=auth
username=123123
password=Asterisk at 123123

[123123]
type=endpoint
context=from-pstn
dtmf_mode=rfc4733
disallow=all
allow=alaw
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
rtp_timeout=60
direct_media=no
trust_id_inbound=no
send_rpid=yes
media_encryption=sdes
inband_progress=no
language=en
auth=123123
outbound_auth=123123
aors=123123

Here when my user registers with UDP and Proxy forward to Asterisk TLS.

    When using pjsip with rewrite contact the IP is updated with Proxy IP instead of local and calls forwarding to Proxy but doing the forward it also rewrites context header protocol from UDP to TLS. Hence my calls start fail.

I need to rewrite the Contaxt here IP only instead of IP and protocol

Below is the sample log for pjsip which not working fine

<— Received SIP request (684 bytes) from TLS:10.10.10.15:45906 —>
REGISTER sip:10.10.10.10;transport=UDP SIP/2.0
Via: SIP/2.0/TLS 10.10.10.15:5061;branch=z9hG4bKe855.a0e258ca3de5139adf1781ce919e86a7.0
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—a6f797de0f863346
Max-Forwards: 69
To: sip:658906 at 10.10.10.10;transport=UDP
From: sip:658906 at 10.10.10.10;transport=UDP;tag=9d10e428
Call-ID: k4sHrL9AjlELoQ_U6d3mIQ…
CSeq: 1 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.9 v2.10.17.3
Allow-Events: presence, kpml, talk
Content-Length: 0
P-Hint: outbound

<— Transmitting SIP response (613 bytes) to TLS:10.10.10.15:45906 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 10.10.10.15:5061;rport=45906;received=10.10.10.15;branch=z9hG4bKe855.a0e258ca3de5139adf1781ce919e86a7.0
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—a6f797de0f863346
Call-ID: k4sHrL9AjlELoQ_U6d3mIQ…
From: sip:658906 at 10.10.10.10;tag=9d10e428
To: sip:658906 at 10.10.10.10;tag=z9hG4bKe855.a0e258ca3de5139adf1781ce919e86a7.0
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1645773968/5a1e2515a15ae40c7f982d2cc372f1f1”,opaque=“73421f31588505a4”,algorithm=md5,qop=“auth”
Server: asterisk Telephony
Content-Length: 0

<— Received SIP request (976 bytes) from TLS:10.10.10.15:45906 —>
REGISTER sip:10.10.10.10;transport=UDP SIP/2.0
Via: SIP/2.0/TLS 10.10.10.15:5061;branch=z9hG4bKb855.dc9dc6093e30fd984f39801c2d2208ac.0
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—d77008763cf147af
Max-Forwards: 69
Contact: sip:658906 at 10.4.6.119:63731;rinstance=d0aef387ddba6ff0;transport=UDP
To: sip:658906 at 10.10.10.10;transport=UDP
From: sip:658906 at 10.10.10.10;transport=UDP;tag=9d10e428
Call-ID: k4sHrL9AjlELoQ_U6d3mIQ…
CSeq: 2 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.9 v2.10.17.3
Authorization: Digest username=“658906”,realm=“asterisk”,nonce=“1645773968/5a1e2515a15ae40c7f982d2cc372f1f1”,uri=“sip:10.10.10.10;transport=UDP”,response=“74ae03622bd7f37b16f1d39da0dd1b75”,cnonce=“f4b85192b0b3d5a91c7f0960f1d1d3a6”,nc=00000001,qop=auth,algorithm=md5,opaque=“73421f31588505a4”
Allow-Events: presence, kpml, talk
Content-Length: 0
P-Hint: outbound

-- Added contact 'sip:658906 at 10.10.10.15:45906;transport=TLS;rinstance=d0aef387ddba6ff0;x-ast-orig-host=10.4.6.119:63731' to AOR '658906' with expiration of 60 seconds

<— Transmitting SIP response (599 bytes) to TLS:10.10.10.15:45906 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.10.10.15:5061;rport=45906;received=10.10.10.15;branch=z9hG4bKb855.dc9dc6093e30fd984f39801c2d2208ac.0
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—d77008763cf147af
Call-ID: k4sHrL9AjlELoQ_U6d3mIQ…
From: sip:658906 at 10.10.10.10;tag=9d10e428
To: sip:658906 at 10.10.10.10;tag=z9hG4bKb855.dc9dc6093e30fd984f39801c2d2208ac.0
CSeq: 2 REGISTER
Date: Fri, 25 Feb 2022 07:26:08 GMT
Contact: sip:658906 at 10.4.6.119:63731**;transport=TLS**;rinstance=d0aef387ddba6ff0;expires=59
Expires: 60
Server: asterisk Telephony
Content-Length: 0

Sample log for SIP using - which is working

<— SIP read from TLS:10.10.10.15:47536 —>
REGISTER sip:10.10.10.10;transport=UDP SIP/2.0
Via: SIP/2.0/TLS 10.10.10.15:5061;branch=z9hG4bK78fe.05d270ab59ab920ff7f0a87c0190f69e.0
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—6161c8385bf65a62
Max-Forwards: 69
Contact: sip:658906 at 10.4.6.119:63731;rinstance=1dce818979687c41;transport=UDP
To: sip:658906 at 10.10.10.10;transport=UDP
From: sip:658906 at 10.10.10.10;transport=UDP;tag=fb43864c
Call-ID: 3R4eHBbQdwxlJV_3gdJI5A…
CSeq: 1 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.9 v2.10.17.3
Allow-Events: presence, kpml, talk
Content-Length: 0
P-Hint: outbound

<------------->
— (15 headers 0 lines) —
Sending to 10.10.10.15:47536 (NAT)
Sending to 10.10.10.15:47536 (NAT)

<— Transmitting (NAT) to 10.10.10.15:47536 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 10.10.10.15:5061;branch=z9hG4bK78fe.05d270ab59ab920ff7f0a87c0190f69e.0;received=10.10.10.15;rport=47536
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—6161c8385bf65a62
From: sip:658906 at 10.10.10.10;transport=UDP;tag=fb43864c
To: sip:658906 at 10.10.10.10;transport=UDP;tag=as11404aa6
Call-ID: 3R4eHBbQdwxlJV_3gdJI5A…
CSeq: 1 REGISTER
Server: Phonon Telephony
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“AWSPOC-1”, nonce=“03f0f566”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3R4eHBbQdwxlJV_3gdJI5A…’ in 32000 ms (Method: REGISTER)

<— SIP read from TLS:10.10.10.15:47536 —>
REGISTER sip:10.10.10.10;transport=UDP SIP/2.0
Via: SIP/2.0/TLS 10.10.10.15:5061;branch=z9hG4bK48fe.6b5f9894323411723b4bb16791edbb82.0
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—4216ee0f07b01a37
Max-Forwards: 69
Contact: sip:658906 at 10.4.6.119:63731;rinstance=1dce818979687c41;transport=UDP
To: sip:658906 at 10.10.10.10;transport=UDP
From: sip:658906 at 10.10.10.10;transport=UDP;tag=fb43864c
Call-ID: 3R4eHBbQdwxlJV_3gdJI5A…
CSeq: 2 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.9 v2.10.17.3
Authorization: Digest username=“658906”,realm=“AWSPOC-1”,nonce=“03f0f566”,uri=“sip:10.10.10.10;transport=UDP”,response=“f2f86257be9d891528d93784e913b68e”,algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 0
P-Hint: outbound

<------------->
— (16 headers 0 lines) —
Sending to 10.10.10.15:47536 (NAT)
– Registered SIP ‘658906’ at 10.10.10.15:47536

<— Transmitting (NAT) to 10.10.10.15:47536 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.10.10.15:5061;branch=z9hG4bK48fe.6b5f9894323411723b4bb16791edbb82.0;received=10.10.10.15;rport=47536
Via: SIP/2.0/UDP 10.4.6.119:63731;rport=63731;branch=z9hG4bK-524287-1—4216ee0f07b01a37
From: sip:658906 at 10.10.10.10;transport=UDP;tag=fb43864c
To: sip:658906 at 10.10.10.10;transport=UDP;tag=as11404aa6
Call-ID: 3R4eHBbQdwxlJV_3gdJI5A…
CSeq: 2 REGISTER
Server: Phonon Telephony
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:658906 at 10.4.6.119:63731;rinstance=1dce818979687c41;transport=UDP;expires=60
Date: Fri, 25 Feb 2022 07:34:14 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3R4eHBbQdwxlJV_3gdJI5A…’ in 32000 ms (Method: REGISTER)

is there an option where we can register pjsip with the contact header of public IP without rewriting the protocol?



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