Hello,
Thanks Gareth for your reply. I assume you're referring to the first INVITE
in my message, which is from the call that works. I don't know why the sdp
displays only iLBC and speex at that point but the Zoiper client that's
making the call is configured to support gsm, speex, ulaw, alaw, and iLBC,
and the call works fine, audio and all, as the sdp that leaves Asterisk
(thus reaches the called peer) actually contains ulaw, gsm and alaw.
In the failing case Asterisk sends the INVITE via Kamailio to the called
webrtc client, and in this message the rtp profile is m=audio 12902 RTP/AVP
0 3 8 101. Kamailio sends the INVITE to the client, which responds with
488. Kamailio notices this and uses rtpengine to handle the rtp, but: the
client will not accept a second INVITE even though the sdp is correct this
time: the client responds with 482 Loop Detected because the Call-ID is the
same as the previous INVITE it got. This is why I can't handle the rtp
using rtpengine, and here things have already gone wrong. So I need the
INVITE to contain correct sdp when it leaves Asterisk, so sdp conversion
and rtpengine would net be needed. Wonder if there's any way to do that?
cheers,
Olli
2014-12-05 18:53 GMT+02:00 Gareth Blades mailinglist+aster...@dns99.co.uk:
On 05/12/14 16:46, Olli Heiskanen wrote:
INVITE that Asterisk (at port 5070) receives:
PU.BL.IC.IP:5060 PU.BL.IC.IP:5070: SIP, length: 1046
INVITE sip:6...@testers.com;transport=UDP SIP/2.0
Record-Route: sip:PU.BL.IC.IP;lr=on;ftag=41030177
Via: SIP/2.0/UDP
PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0
Via: SIP/2.0/UDP
AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z-
Max-Forwards: 69
Contact: sip:7...@ast.er.isk.ip:38699;transport=UDP
sip:7...@ast.er.isk.ip:38699;transport=UDP
To: sip:6...@testers.com;transport=UDP
From: 771sip:7...@testers.com;transport=UDP;tag=41030177
Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer,
X-cisco-serviceuri
User-Agent: Z 3.2.21357 r21367
Allow-Events: presence, kpml
Content-Length: 239
v=0
o=Z 0 0 IN IP4 AST.ER.ISK.IP
s=Z
c=IN IP4 AST.ER.ISK.IP
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
This client is saying it only supports speex and iLBC and would prefer
them in that order.
Your sip.conf appears to only permit alaw, ulaw and gsm so there is no
mutual supported codec and hence the call fails.
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