[asterisk-users] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work

[Olli Heiskanen]

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+asterisk at 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:660 at 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:771 at AST.ER.ISK.IP:38699;transport=UDP>
> <sip:771 at AST.ER.ISK.IP:38699;transport=UDP>
>  To: <sip:660 at testers.com;transport=UDP>
>  From: "771"<sip:771 at 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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