2 Apr
2018
2 Apr
'18
3:59 p.m.
### 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@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@10.1.1.138>;tag=as7c4d457d\r\n
To: <sip:test2@10.1.1.38:5060>\r\n
Contact: <sip:test@10.1.1.138:5060>\r\n
Call-ID: 2ec6d77f5a56ca0f59ab0d2118d2f7d5@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
<!--
If the issue is exposed by processing specific SIP messages, grab them with ngrep or save
in a pcap file, then add them next, or attach to issue, or provide a link to download them
(e.g., to a pastebin site).
-->
```
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```
### Possible Solutions
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request with a fix.
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### Additional Information
* **Kamailio Version** - output of `kamailio -v`
```
(paste your output here)
```
* **Operating System**:
<!--
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CentOS 7.1, ...), MacOS, xBSD, Solaris, ...;
Kernel details (output of `uname -a`)
-->
```
(paste your output here)
```
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