Failed to Parse SDP Message

[John Forsyth]

Hello,

Thanks in advance for any help.

I am encountering a "Failed to parse sdp" sipml5 error (see below) when using it in combination with webrtc2sip proxy and an embedded SIP server. I am using the webrtc2sip as a proxy to communicate to this embedded SIP server using UDP, as it does not support WebSockets directly.

I am running everything on a local LAN with the follow IP address assignment:

Browser: 192.168.1.2

webrtc2sip Proxy: 192.168.1.56

Embedded SIP server: 192.168.1.32

The versions I am using are as follows:

Browser: Chrome 35.0.1916.153 m

SIPML5 API: 1.4.217

webrtc2sip: 2.6.0

Unfortunately, I do not have any insight into the embedded SIP server. It is running on a device I am attempting to integrate with, and do not have source code, etc. It does work with local (windows) SIP clients that do UDP directly.

I have attached wireshark captures from both the browser PC, and from the system running webrtc2sip proxy. It appears the embedded SIP server is replying, but the browser fails to parse it's SDP, so the SIP handshake breaks at this point.

I have also attached the complete Chrome javascript console log.

I am only trying to achieve audio streaming; video is not required.

Any help is greatly appreciated.

SIPML5 Error message:

Failed to parse sdp message: v=0

o=alice 1404128761 800000 IN IP4 192.168.1.32

s= 

c=IN IP4 192.168.1.32

t=0 0

m=audio 65080 UDP/TLS/RTP/SAVPF 0

c=IN IP4 174.94.136.50

a=rtcp:61199 IN IP4 174.94.136.50

a=candidate:2437072876 1 udp 2122260223 192.168.1.2 61199 typ host generation 0

a=candidate:2437072876 2 udp 2122260223 192.168.1.2 61199 typ host generation 0

a=candidate:941443129 1 udp 1686052607 174.94.136.50 61199 typ srflx raddr 192.168.1.2 rport 61199 generation 0

a=candidate:941443129 2 udp 1686052607 174.94.136.50 61199 typ srflx raddr 192.168.1.2 rport 61199 generation 0

a=candidate:3753982748 1 tcp 1518280447 192.168.1.2 0 typ host generation 0

a=candidate:3753982748 2 tcp 1518280447 192.168.1.2 0 typ host generation 0

a=ice-ufrag:TOTnhm7b3e0vUkCc

a=ice-pwd:GOPd9GnwyHS/+HftecAHzIdd

a=ice-options:google-ice

a=fingerprint:sha-256 9B:FE:A4:87:F6:31:BE:F1:2B:EF:2A:50:45:EC:65:AA:15:FE:AE:E4:CC:57:95:89:91:32:82:D7:FC:3F:04:EA

a=setup:actpass

a=mid:audio

a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level

a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

a=sendrecv

a=rtcp-mux

a=rtpmap:0 PCMU/8000

a=maxptime:60

a=ssrc:1616542168 cname:83QhuYezfWbAvHRU

a=ssrc:1616542168 msid:AhYbwMgsDk2omLqBmZDezgIu4djkcFfDpV17 1f3f1b23-f782-47ba-98a1-d5c69007cfd0

a=ssrc:1616542168 mslabel:AhYbwMgsDk2omLqBmZDezgIu4djkcFfDpV17

a=ssrc:1616542168 label:1f3f1b23-f782-47ba-98a1-d5c69007cfd0

 SIPml-api.js:1

Failed to parse remote sdp message SIPml-api.js:1

State machine: Exec function failed. Moving to the termnial state SIPml-api.js:1

=== INVITE Dialog terminated === SIPml-api.js:1

PeerConnection::stop() SIPml-api.js:1

State machine: Exec function failed. Moving to the termnial state SIPml-api.js:1

The FSM is in the final state SIPml-api.js:1

__tsip_transport_ws_onmessage SIPml-api.js:1

[Mamadou DIOP]

There is an extra 0x00 byte in the 200 from Asterisk. This is probably the problem.

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<SIPML5 Broswer Console.txt><browser-client-capture.pcapng><webrtc2sip-proxy-capture.pcapng>

[John Forsyth]

Thank you for the reply & advice.

Does any mechanism existing, either with SIMPL5, or webrtc2sip, to handle (or remove) the extra 0x00 that is causing the issue? 

I don't have any control over what comes out of this SIP server unfortunately.'

[Yusuf Siddiqui]

Hey John
Webrtc itself has many issues,unfortunately doubango is also growing and may not have answers for your questions.
Doubango:
It would have been better ,if you can have a KB(Knowledge Base) with frequent issues and corresponding answers for the same. 

Regards

Mohd Yusuf Siddiqui
email: yusuf.s...@fiyutech.com

Mob. +91.995.899.521.8

Off:+91.120.437.209.3

U.S. +120.975.347.57

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