26 May
2016
26 May
'16
4:24 p.m.
Hi all,
Im struggling with incoming calls to webrtc clients.
I am trying to change INVITE's SDP using rtpengine from RTP/AVP or
RTP/SAVP to RTP/SAVPF towards my webrtc clients. This works well.
But if I send INVITE with SDP that includes RTP/AVP and RTP/SAVP as
well, rtpengine creates incorrect SDP. It includes twice m=audio
RTP/SAVPF and my WebRTC client does not like this INVITE and forbiddeds
it by 488 Not Acceptable Here.
Example of my incorrect INVITE :
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 1393
v=0
o=SBC 1464238673 1464238674 IN IP4 <MyPublicIP>
s=SBC
c=IN IP4 <MyPublicIP>
t=0 0
m=audio 30454 RTP/SAVPF 102 9 8 0 3 18
a=rtpmap:102 SILK/8000
a=fmtp:102 useinbandfec=1; usedtx=0
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=rtcp:30455
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:5X2wgU83bIwrwkHSymjnw48SiMGa1n+eIl50yr99
a=setup:actpass
a=fingerprint:sha-1
11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88
a=ice-ufrag:iglHZqry
a=ice-pwd:eyVXbFNoviYYlu5uuRZlnixFwQ
a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30454 typ host
a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30455 typ host
m=audio 30484 RTP/SAVPF 102 9 8 0 3 18
a=rtpmap:102 SILK/8000
a=fmtp:102 useinbandfec=1; usedtx=0
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=ptime:20
a=sendrecv
a=rtcp:30485
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:ozDJ3G/wmaYedQbcTafbhaTIt6raIJa6ugrLUC99
a=setup:actpass
a=fingerprint:sha-1
11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:88
a=ice-ufrag:gM1viWqu
a=ice-pwd:zdS4tP1Mj4hLTe6huBWYrhtI1s
a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <MyPublicIP> 30484 typ host
a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <MyPublicIP> 30485 typ host
My setup :
incoming call -> MediaServer (freeswitch) -> SIP & Websocket proxy
(Kamailio + Rtpengine) -> webRTC clients
Is there anybody who could help me with this issue or explain me how to
remove the second line of m=audio RTP/SAVPF ?
--
Jan