<div dir="ltr">Hello,<div><br></div><div>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.</div><div><br></div><div>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?</div><div><br></div><div>cheers,</div><div>Olli</div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2014-12-05 18:53 GMT+02:00 Gareth Blades <span dir="ltr"><<a href="mailto:mailinglist+asterisk@dns99.co.uk" target="_blank">mailinglist+asterisk@dns99.co.uk</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"><div><div class="h5">
On 05/12/14 16:46, Olli Heiskanen wrote:<br>
<blockquote type="cite">
<div>INVITE that Asterisk (at port 5070) receives:</div>
<div>PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046</div>
<div><span style="white-space:pre-wrap"> </span>INVITE <a href="mailto:sip%3A660@testers.com" target="_blank">sip:660@testers.com</a>;transport=UDP
SIP/2.0</div>
<div><span style="white-space:pre-wrap"> </span>Record-Route:
<sip:PU.BL.IC.IP;lr=on;ftag=41030177></div>
<div><span style="white-space:pre-wrap"> </span>Via:
SIP/2.0/UDP
PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0</div>
<div><span style="white-space:pre-wrap"> </span>Via:
SIP/2.0/UDP
AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z-</div>
<div><span style="white-space:pre-wrap"> </span>Max-Forwards:
69</div>
<div><span style="white-space:pre-wrap"> </span>Contact:
<a href="mailto:sip:771@AST.ER.ISK.IP:38699;transport=UDP" target="_blank"><sip:771@AST.ER.ISK.IP:38699;transport=UDP></a></div>
<div><span style="white-space:pre-wrap"> </span>To: <<a href="mailto:sip%3A660@testers.com" target="_blank">sip:660@testers.com</a>;transport=UDP></div>
<div><span style="white-space:pre-wrap"> </span>From:
"771"<<a href="mailto:sip%3A771@testers.com" target="_blank">sip:771@testers.com</a>;transport=UDP>;tag=41030177</div>
<div><span style="white-space:pre-wrap"> </span>Call-ID:
YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk.</div>
<div><span style="white-space:pre-wrap"> </span>CSeq: 2
INVITE</div>
<div><span style="white-space:pre-wrap"> </span>Allow:
INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE</div>
<div><span style="white-space:pre-wrap"> </span>Content-Type:
application/sdp</div>
<div><span style="white-space:pre-wrap"> </span>Supported:
replaces, norefersub, extended-refer, timer, X-cisco-serviceuri</div>
<div><span style="white-space:pre-wrap"> </span>User-Agent: Z
3.2.21357 r21367</div>
<div><span style="white-space:pre-wrap"> </span>Allow-Events:
presence, kpml</div>
<div><span style="white-space:pre-wrap"> </span>Content-Length:
239</div>
<div><br>
</div>
<div><span style="white-space:pre-wrap"> </span>v=0</div>
<div><span style="white-space:pre-wrap"> </span>o=Z 0 0 IN
IP4 AST.ER.ISK.IP</div>
<div><span style="white-space:pre-wrap"> </span>s=Z</div>
<div><span style="white-space:pre-wrap"> </span>c=IN IP4
AST.ER.ISK.IP</div>
<div><span style="white-space:pre-wrap"> </span>t=0 0</div>
<div><span style="white-space:pre-wrap"> </span>m=audio 8000
RTP/AVP 3 110 8 0 98 101</div>
<div><span style="white-space:pre-wrap"> </span>a=rtpmap:110
speex/8000</div>
<div><span style="white-space:pre-wrap"> </span>a=rtpmap:98
iLBC/8000</div>
<div><span style="white-space:pre-wrap"> </span>a=fmtp:98
mode=20</div>
<div><span style="white-space:pre-wrap"> </span>a=rtpmap:101
telephone-event/8000</div>
<div><span style="white-space:pre-wrap"> </span>a=fmtp:101
0-15</div>
<div><span style="white-space:pre-wrap"> </span>a=sendrecv</div>
</blockquote>
<br></div></div>
This client is saying it only supports speex and iLBC and would
prefer them in that order.<br>
Your sip.conf appears to only permit alaw, ulaw and gsm so there is
no mutual supported codec and hence the call fails.<br>
<br>
</div>
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