[Spce-user] SDP issues-SAVP

[H Yavari]

Hi,
I did another test: calls with this flow : Extension--->Asterisk---->NGCP---->Peer (Asterisk)--->extension are established but without audio and terminated after rtp timeout expiry.I use "prefer SRTP and force avp" in ngcp domain configurations and "Fore RTP and Force AVP" in peer server configuration's.

In rtp.log : 
Mar 16 13:57:44 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30741] Successful STUN binding request from 192.168.1.122:22629
Mar 16 13:57:44 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30741] STUN: using this candidate
Mar 16 13:57:44 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30740] SRTP output wanted, but no crypto suite was negotiated
Mar 16 13:57:48 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30740] Confirmed peer address as 192.168.1.122:22628
Mar 16 13:57:49 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30741] SRTCP output wanted, but no crypto suite was negotiated
Mar 16 13:57:49 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30741] Confirmed peer address as 192.168.1.122:22629
Mar 16 13:57:59 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30740] SRTP output wanted, but no crypto suite was negotiated
Mar 16 13:58:04 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30741] SRTCP output wanted, but no crypto suite was negotiated
Mar 16 13:58:14 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30740] SRTP output wanted, but no crypto suite was negotiated
Mar 16 13:58:19 spce rtpengine[6966]: [6fde00ca3efba05e454b4014006ec98c at 192.168.1.133 port 30741] SRTCP output wanted, but no crypto suite was negotiated

I'm using version 3.7.2 .
How can I solve this issue?
Regards.

     From: H Yavari <hyavari at rocketmail.com>
Hi,
I have ngcp with an asterisk peer. My Asterisk and NGCP have their calls (webrtc or usual calls) and every things is Ok.But when I call from asterisk extension to ngcp extension, calls will be terminated immediately after answer . I see this error in the Asterisk side:
"Invalid syntax in RTP audio format list: media type>"
The SDP part of "OK" message from NGCP is :
 
v=0
o=Mozilla-SIPUA-36.0.1 9775 0 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-lite
m=audio 30050 RTP/SAVP media type>
c=IN IP4 0.0.0.0
a=direction:both
a=sendonly
a=rtcp:30051
a=setup:active
a=fingerprint:sha-1 6E:66:B1:CF:06:6A:19:FA:FC:46:FA:BA:25:DC:5F:55:89:9C:A6:8A
a=ice-ufrag:8tdif51p
a=ice-pwd:xij0qJh8ArQFPyDdUc01gT7omkFA
a=candidate:ssNViQ3fGqIAIUIO 1 UDP 2130706431 0.0.0.0 30050 typ host
a=candidate:ssNViQ3fGqIAIUIO 2 UDP 2130706430 0.0.0.0 30051 typ host
I use "prefer SRTP and force avp" in ngcp domain configurations.
So what is the problem? plz help.

B.R,
H.Yavari




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