Olli Heiskanen
2014-Dec-05 16:46 UTC
[asterisk-users] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
Hello,
I'd appreciate your comments on the following problem I'm having, please
forgive me if this is something obvious, I've been scratching my head on
this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls
from outside. I have both webrtc and sip clients, where webrtc peers are
defined according to sip.js instructions (
http://sipjs.com/guides/server-configuration/asterisk/). Calls between
these work nicely without problems. Now when I call from outside, from an
external Asterisk 11.5 server, I end up having problems calling from a sip
client to a webrtc client. The Asterisk I have on my main testing server is
the latest current 11.14.1.
When there's an internal call, Asterisk changes the sdp in the INVITE
message and handles the rtp nicely, but it does not do so when the call
comes from outside. Why not? Instead, it sends back 488 Not acceptable
here. If I react to that in Kamailio and use rtpengine to rewrite the sdp,
Asterisk accepts the INVITE and sends it to the websocket peer, but the sdp
contains a very simple sdp with RTP/AVP profile. This I'd consider invalid
behavior, since Asterisk knows the called party is webrtc and the INVITE
already contains valid sdp with RTP/SAVPF profile. It's likely I have
something wrong in my setup, or maybe I've overlooked something relevant?
Question is, what is causing this behavior and what can I do to fix it?
Either I'd need Asterisk to handle the sdp and rtp like it does for
internal calls (which would be preferable in this case) or after the 488
sent by Asterisk I'd need Asterisk to relay the sdp offered by
Kamailio/rtpengine as such without rewriting it.
Here the call works fine (internal call from sip peer 771 to webrtc peer
660):
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>
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
Invite that Asterisk sends:
PU.BL.IC.IP:5070 > PU.BL.IC.IP:5060: SIP, length: 1238
INVITE sip:660 at PU.BL.IC.IP:5060 SIP/2.0
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK26a2386a;rport
Max-Forwards: 70
From: "771 win8 minipc" <sip:771 at
testers.com:5070>;tag=as05e60cc6
To: <sip:660 at PU.BL.IC.IP:5060>
Contact: <sip:771 at PU.BL.IC.IP:5070>
Call-ID: 7985f7161fcf1a6824b8388d451bec16 at testers.com
CSeq: 102 INVITE
User-Agent: I Am the Devil
Date: Fri, 05 Dec 2014 15:50:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 663
v=0
o=root 777617621 777617621 IN IP4 PU.BL.IC.IP
s=Asterisk PBX 11.14.1
c=IN IP4 PU.BL.IC.IP
t=0 0
m=audio 15662 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:75a7e84431d15f682bd728ee10bd867d
a=ice-pwd:028c19574216643c12188a8530f278f8
a=candidate:H5bdd423d 1 UDP 2130706431 PU.BL.IC.IP 15662 typ host
a=candidate:H5bdd423d 2 UDP 2130706430 PU.BL.IC.IP 15663 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05
a=sendrecv
Here the call fails (sip peer 201 calls from outside the server to webrtc
peer 660):
Invite that Asterisk receives:
PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1345
INVITE sip:660%40testers.com at PU.BL.IC.IP SIP/2.0
Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=as4647f03c;nat=yes>
Via: SIP/2.0/UDP
PU.BL.IC.IP;branch=z9hG4bK3264.8a896801756c527f2496fdc14e3f30ad.0
Via: SIP/2.0/UDP 192.168.0.201:5060
;rport=5060;received=AST.ER.ISK.IP;branch=z9hG4bK56f5698e
Max-Forwards: 69
From: "Pirjo Ahvenainen" <sip:201 at
192.168.0.201>;tag=as4647f03c
To: <sip:660%40testers.com at PU.BL.IC.IP>
Contact: <sip:201 at 192.168.0.201:5060;alias=AST.ER.ISK.IP~5060~1>
Call-ID: 69e66f05330de0063b5eba760191da6c at 192.168.0.201:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.0
Date: Tue, 02 Dec 2014 08:34:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 547
v=0
o=root 1854132825 1854132825 IN IP4 PU.BL.IC.IP
s=Asterisk PBX 11.5.0
c=IN IP4 PU.BL.IC.IP
t=0 0
a=ice-lite
m=audio 12516 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=rtcp:12517
a=ice-ufrag:5vj2cfWg
a=ice-pwd:OdPg0e2qmExbbvAXUTLT3NI8g28s
a=candidate:dEk0op4jY8ZkdPXr 1 UDP 2130706431 PU.BL.IC.IP 12516 typ host
a=candidate:dEk0op4jY8ZkdPXr 2 UDP 2130706430 PU.BL.IC.IP 12517 typ host
And the INVITE the Asterisk sends:
PU.BL.IC.IP:5070 > PU.BL.IC.IP:5060: SIP, length: 847
INVITE sip:660 at testers.com SIP/2.0
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK6f718231
Max-Forwards: 70
From: "771 win8 minipc" <sip:201 at
PU.BL.IC.IP:5070>;tag=as2931af14
To: <sip:660 at testers.com>
Contact: <sip:201 at PU.BL.IC.IP:5070>
Call-ID: 04a3975e3bc84e6e32bfdc1905791913 at PU.BL.IC.IP:5070
CSeq: 102 INVITE
User-Agent: I Am the Devil
Date: Fri, 05 Dec 2014 15:52:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=root 1272725383 1272725383 IN IP4 PU.BL.IC.IP
s=Asterisk PBX 11.14.1
c=IN IP4 PU.BL.IC.IP
t=0 0
m=audio 11906 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
And how are my peers configured? The calling peer on the separate Asterisk
server is configured quickly in sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
tcpbindaddr = 0.0.0.0
tcpenable = yes
limitonpeers = yes
rtcachefriends = no
tos_sip = cs3
tos_audio = ef
[201]
type = friend
secret = myverysecretpassword
context = mytestcontext
callerid = "201 User" <201>
host = dynamic
port = 5060
disallow = all
allow = alaw
allow = ulaw
allow = gsm
qualify = yes
nat = no
canreinvite = no
and the dial command is like this:
exten => 666,n,Dial(SIP/PU.BL.IC.IP/660 at testers.com)
And the called number 660 is in the realtime sippeers table:
id: 4
type: friend
name: 660
host: dynamic
secret: NULL
encryption: yes
avpf: yes
icesupport: yes
ipaddr: PU.BL.IC.IP
port: 5060
regseconds: 1417782175
defaultuser: 660
fullcontact: sip:660 at PU.BL.IC.IP:5060
lastms: 0
useragent:
context: mytestnumbercontext
directmedia: no
deny: 0.0.0.0/0.0.0.0
permit: PU.BL.IC.IP
nat: force_rport,comedia
transport: ws,wss,udp
language: NULL
disallow: NULL
allow: NULL
force_avp: yes
callerid: 660 test
amaflags: NULL
mailbox: NULL
regexten: NULL
regserver:
fromdomain: testers.com
videosupport: no
contactpermit: NULL
contactdeny: NULL
fullname: 660 win8
subscribemwi: NULL
dtlsenable: yes
dtlsverify: no
dtlscertfile: /etc/asterisk/keys/asterisk.pem
dtlsprivatekey: /etc/asterisk/keys/asterisk.pem
dtlssetup: actpass
sippasswd: a84a4ddcda13d13c9573d5294047b6a2
rpid: NULL
domain: testers.com
sippasswd2: 5c4671ae1043e6116118fed39bee091a
callbackextension: NULL
insecure: NULL
cheers,
Olli
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Gareth Blades
2014-Dec-05 16:53 UTC
[asterisk-users] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
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 > <mailto:sip%3A660 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> > To: <sip:660 at testers.com <mailto:sip%3A660 at testers.com>;transport=UDP> > From: "771"<sip:771 at testers.com > <mailto:sip%3A771 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=sendrecvThis 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141205/f2a38d33/attachment.html>
Olli Heiskanen
2014-Dec-05 19:03 UTC
[asterisk-users] Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
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. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141205/a3685c6d/attachment.html>