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altanai / sipml5

Automatically exported from code.google.com/p/sipml5
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Could not set SRTP policies #83

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?

1. Connect from Software SIP Phone (I have tried both X-Lite and MicroSIP(PJSIP)
2. Upon answering the call in either software SIP or sipML5 demo the Asterix 
server fails with two similar warnings and then a third other warning regarding 
SRTP Policies and a rejection of the secure audiostream. (See below).

What is the expected output? What do you see instead?

I expect the call to connect and the VOIP call to go through. It fails and the 
call is dropped upon answering.

What version of the product are you using? On what operating system?

Asterix installed as per guide from SVN and patched as per the instructions on 
the sipml5 WIKI.

Please provide any additional information below.

Regardless how wich way i try to call i get this from a soft SIP Phone 
MicroSIP(PJSIP) to sipML5 phone demo in Chrome:

== WebSocket? connection from '10.1.1.173:56439' for protocol 'sip' accepted 
using version '13'
-- Registered SIP '1060' at 10.1.1.173:56439
> Saved useragent "IM-client/OMA1.0 sipML5-v1.2013.03.27" for peer 1060
== Using SIP RTP CoS mark 5
-- Executing [1060@default:1] Gosub("SIP/1061-00000000", 
"1060,stdexten(SIP/1060)") in new stack -- Executing [1060@default:50000] 
NoOp?("SIP/1061-00000000", "Start stdexten") in new stack -- Executing 
[1060@default:50001] Set("SIP/1061-00000000", "LOCAL(ext)=1060") in new stack 
-- Executing [1060@default:50002] Set("SIP/1061-00000000", 
"LOCAL(dev)=SIP/1060") in new stack -- Executing [1060@default:50003] 
Set("SIP/1061-00000000", "LOCAL(cntx)=") in new stack -- Executing 
[1060@default:50004] Set("SIP/1061-00000000", "LOCAL(mbx)=1060") in new stack 
-- Executing [1060@default:50005] Dial("SIP/1061-00000000", "SIP/1060,20") in 
new stack
== Using SIP RTP CoS mark 5
-- Called SIP/1060 -- SIP/1060-00000001 is ringing
2 20:42:10? WARNING6658?[C-00000001]: sip/sdp_crypto.c:173 sdp_crypto_activate: 
Could not set SRTP policies 2 20:42:10? WARNING6658?[C-00000001]: 
chan_sip.c:10421 process_sdp: Rejecting secure audio stream without encryption 
details: audio 57642 RTP/SAVPF 0 8 101

-- SIP/1060-00000001 answered SIP/1061-00000000
== Spawn extension (default, 1060, 50005) exited non-zero on 'SIP/1061-00000000'
Expert settings is set to: Video = DISABLED RTCWeb BREAKER = ENABLED WebSocket? 
Server URL = ws://10.1.1.171:8088/ws WebProxy? = empty Ice = empty

i can see the call coming in but when i answer it the call is dropped i don't 
know if it's something with the encryption of the call or what is causing it. I 
could also use some pointers on this.

Best regards and thanks for a cool project //Troels

Original issue reported on code.google.com by troe...@gmail.com on 2 Apr 2013 at 7:48

GoogleCodeExporter commented 9 years ago 
Added some logs, configs and ss of sipML5 setup.

Original comment by troe...@gmail.com on 2 Apr 2013 at 7:58

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GoogleCodeExporter commented 9 years ago

Original comment by troe...@gmail.com on 2 Apr 2013 at 8:00

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GoogleCodeExporter commented 9 years ago 
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GoogleCodeExporter commented 9 years ago 
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GoogleCodeExporter commented 9 years ago 
Could not set SRTP policies
----------------------------

This is not a Sipml5 issue, but looks come from the media-level SDP subsystem 
(audio) in asterisk.

I have compiled asterisk version 1.8, 11.5, 11.6 11.7 and 12+pjproject (beta), 
using 
./configure --with-crypto --with-ssl=ssl --with-srtp 

All of them got the same behaviour. Even, after to apply a number of patches 
recommended in this website. 

Asterisk version 11.5, 11.6 and 11.7 don't understand "a=candidates" lines 
reported by the WebRTC client (STUN protocol). As a result

1) res_srtp.c: Adding new policy for SSRC
2) sip/sdp_crypto.c: SRTP policy activated
3) sip/sdp_crypto.c: Accepting crypto tag 0

are not processed and "Could not set SRTP policies"

Asterisk 12 can process a=candidates, but still show the same error "Could not 
set SRTP policies" 

BTW: module res_srtp was always loaded at the start without errors.

The only asterisk version I have detected which set up the policy is asterisk 
11.4 (package).

I could be wrong and I should to provide more parameters after 
./configure --XXX -YYY 
but I was unable find what is missing. 

Miguel Oyarzo,
Melbourne

Original comment by miguel.o...@medulla.com.au on 23 Nov 2013 at 9:01