[sr-dev] [kamailio/kamailio] Double SPACE after in the VIA WSS (#1491)

[ovoshlook]

### Description

Hi. Seems lime that after WSS in the VIA kamailio adds double SPACE
It makes kamailio incompatable with jssip.net (tried 5.1.2 and 5.0.6 versions) 

Looks like this happens with advertise address added on the interface

#### Reproduction

Just connect to kamailio via WSS with advertized address and try to make incoming call into WSS client

#### Debugging Data

I just put this link to JsSIP.net here
https://groups.google.com/forum/#!topic/jssip/2lDyqgvZgrY

Also here is a string of my VIA header that gives issue

SIP/2.0/WSS ​1.2.3.4:5061;branch=z9hG4bKd302.4c862421515641e7b59b9f3ba7f8eab4.1
Thant can be checked here
https://www.textmagic.com/free-tools/unicode-detector

Log INVITE from the browser (only added \r\n at the end of the each string because of notepad++ not shows it after copying):

INVITE sip:2hsq8ob4 at daotd6p7hil4.invalid;transport=ws SIP/2.0\r\n
Record-Route: <sip:1.2.3.4:5061;transport=ws;r2=on;lr;ftag=as7c4d457d;nat=yes>\r\n
Record-Route: <sip:1.2.3.4;r2=on;lr;ftag=as7c4d457d;nat=yes>\r\n
Via: SIP/2.0/WSS ​1.2.3.4:5061;branch=z9hG4bK110e.476a7e8aa7db8de3a4d7b01aed1e7ef5.1\r\n
Via: SIP/2.0/UDP 10.1.1.138:5060;rport=5060;branch=z9hG4bK7122bcfb\r\n
Max-Forwards: 69\r\n
From: "test" <sip:test at 10.1.1.138>;tag=as7c4d457d\r\n
To: <sip:test2 at 10.1.1.38:5060>\r\n
Contact: <sip:test at 10.1.1.138:5060>\r\n
Call-ID: 2ec6d77f5a56ca0f59ab0d2118d2f7d5 at 10.1.1.138:5060\r\n
CSeq: 102 INVITE\r\n
User-Agent: Asterisk PBX 15.3.0\r\n
Date: Mon, 02 Apr 2018 10:30:57 GMT\r\n
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE\r\n
Supported: replaces, timer\r\n
Content-Type: application/sdp\r\n
Content-Length: 642\r\n
\r\n
v=0\r\n
o=root 1538180518 1538180518 IN IP4 1.2.3.4\r\n
s=Asterisk PBX 15.3.0\r\n
c=IN IP4 1.2.3.4\r\n
t=0 0\r\n
a=group:BUNDLE audio\r\n
m=audio 31858 RTP/SAVPF 8 0 101\r\n
a=maxptime:150\r\n
a=rtpmap:8 PCMA/8000\r\n
a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=fmtp:101 0-16\r\n
a=sendrecv\r\n
a=rtcp:31859\r\n
a=rtcp-mux\r\n
a=setup:actpass\r\n
a=mid:audio\r\n
a=fingerprint:sha-1 95:6F:40:F2:76:B3:E3:1E:DA:29:04:60:F1:F7:0A:DA:5E:D4:67:9F\r\n
a=ice-ufrag:eSuU9ztd\r\n
a=ice-pwd:j693abw9QWiaX81TlbvyHBD8FU\r\n
a=candidate:5Qp8pvPCHLmm6iUU 1 UDP 2130706431 1.2.3.4 31858 typ host\r\n
a=candidate:5Qp8pvPCHLmm6iUU 2 UDP 2130706430 1.2.3.4 31859 typ host\r\n


Actually i did not found any information about is it MUST be only one sace at the Grammar of SIP message based on RFC 2234 but I may be wrong.
```

#### SIP Traffic

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```
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```

### Possible Solutions

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### Additional Information

  * **Kamailio Version** - output of `kamailio -v`

```
(paste your output here)
```

* **Operating System**:

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Kernel details (output of `uname -a`)
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```
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```


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