Hello 
I have a problem with my Asterisk - WebRTC Configuration.
I want to make an audio call from one sipml5 client to an other one. I can hear both sides. So far so good.
But I noticed that Asterisk changes the complete SIP Invite Message. As you can see it changes everything, Audio-Codecs, Crypto…
I’m a little bit confused is this normal??
Thanks in advance for your help 
Edit: I took this tutorial http://highsecurity.blogspot.de/2012/12/webrtc-and-asterisk-11-using-sipml5.htmlto install everything.
This is the Invite Message from Browser 1 to Asterisk:
INVITE sip:8001@192.168.56.101 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKUSPMcGaC5ga0Z8rwmpU57C2RDgAn2llh;rport
From: "8000"sip:8000@192.168.56.101;tag=6fRzG1QEKnGPiq3UCtH9
To: sip:8001@192.168.56.101
Contact: "8000"sip:8000@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language=“en,fr"
Call-ID: f431c29b-bf2f-b1e9-3576-cf74d48123ad
CSeq: 62090 INVITE
Content-Type: application/sdp
Content-Length: 1510
Max-Forwards: 70
Authorization: Digest username=“8000”,realm=“192.168.56.101”,nonce=“15556c4a”,uri="sip:8001@192.168.56.101”,response=“46a8e1c918d23c74603070c1b163cdad”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B
Organization: Doubango Telecom
v=0
o=- 101739073460546430 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS N7A3Pw1XmI5WaKcGebWZyi3agyqYlpKPdM1v
m=audio 50853 RTP/SAVPF 111 103 104 0 8 107 106 105 13 126
c=IN IP4 192.168.56.1
a=rtcp:50853 IN IP4 192.168.56.1
a=candidate:2999745851 1 udp 2113937151 192.168.56.1 50853 typ host generation 0
a=candidate:2999745851 2 udp 2113937151 192.168.56.1 50853 typ host generation 0
a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=candidate:4233069003 2 tcp 1509957375 192.168.56.1 0 typ host generation 0
a=ice-ufrag:g8yGjswVjqqPXV2m
a=ice-pwd:3WN3tN5I8aZrkdrsT0gG+VjO
a=ice-options:google-ice
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:x8Y3lZbZNIY9hWkhrjDjj/CVsX7w9Qo9RvQt6bdh
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:+Qy3bKX19gT9oJy667Zg41qpf43iz0PlQGkzR7Jv
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:4136192023 cname:bKNVSpLEpdaSayCa
a=ssrc:4136192023 msid:N7A3Pw1XmI5WaKcGebWZyi3agyqYlpKPdM1v N7A3Pw1XmI5WaKcGebWZyi3agyqYlpKPdM1va0
a=ssrc:4136192023 mslabel:N7A3Pw1XmI5WaKcGebWZyi3agyqYlpKPdM1v
a=ssrc:4136192023 label:N7A3Pw1XmI5WaKcGebWZyi3agyqYlpKPdM1va0
And this one from Asterisk to Browser 2:
INVITE sip:8001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 192.168.56.101:5060;branch=z9hG4bK252faa37;rport
Max-Forwards: 70
From: “Test One” sip:8000@192.168.56.101;tag=as61deb75a
To: sip:8001@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=ws
Contact: sip:8000@192.168.56.101:5060;transport=WS
Call-ID: 2e4433da4b7818e671e466f9285fa841@192.168.56.101:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.1
Date: Tue, 05 Nov 2013 13:52:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 600
v=0
o=root 307719493 307719493 IN IP4 192.168.56.101
s=Asterisk PBX 11.5.1
c=IN IP4 192.168.56.101
t=0 0
m=audio 19892 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:57516c7328ce8d2e20635a3253228145
a=ice-pwd:78ace80d0bae8fae410f341c75f8bac9
a=candidate:Hc0a83865 1 UDP 2130706431 192.168.56.101 19892 typ host
a=candidate:Hc0a83865 2 UDP 2130706430 192.168.56.101 19893 typ host
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrxS1lKltTclC0VOQACJaJEHHamyRn+00Tjttdj/