Olli Heiskanen
2014-Jul-26 09:58 UTC
[asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Greetings,
I've noticed a problem that might originate from my Asterisk configuration,
could use a hand in sorting it out. Problem is a 488 response from Asterisk
whenever it gets RTP/SAVPF profile in the SDP.
My current setup has Asterisk Kamailio realtime integration, and Kamailio
uses dispatcher to route calls for Asterisk to handle. Now I have only one
Asterisk, on the same machine as Kamailio. The version is 11.10.2. With
Kamailio I use rtpengine, which affects SDP descriptions when 488 response
is received.
My goal is to enable two websocket clients using Chrome to call each other,
using Kamailio as outbound proxy. Kamailio routes signaling to Asterisk,
and then back to clients. Currently the problem is RTP, when INVITE is
received from client A to Kamailio, it is relayed to Asterisk. Asterisk
responds with 488 Not Acceptable here and the cli says:
NOTICE[11642][C-00000006]: chan_sip.c:10124 process_sdp: Received SAVPF
profle in audio offer but AVPF is not enabled, enabling: audio 30212
RTP/SAVPF 111 103 104 0 8 106 105 13 126
WARNING[11642][C-00000006]: chan_sip.c:10509 process_sdp: Rejecting secure
audio stream without encryption details: audio 30212 RTP/SAVPF 111 103 104
0 8 106 105 13 126
Strange thing is, I don't know why Asterisk says AVPF is not enabled. The
warning about rejecting the audio stream must be behind the 488 response
but I didn't find any answers that would solve my case so I must turn to
you guys. In my sip.conf I have savpf=yes, but is there something else I
need to enable or change in the configs or change my peer configurations?
I'm not sure if this is relevant but I checked that Asterisk was
successfully compiled with res_srtp module.
Here's my sip.conf contents:
bindport = 5070 ; using this since Kamailio is at 5060
bindaddr = PU.BL.IC.IP
tcpenable = yes ;no
limitonpeers = yes
rtcachefriends = yes ; for realtime
rtupdate=yes
tos_sip=cs3
tos_audio=ef
useragent=MyAsterisk
realm = myrealm.com
autodomain=no
domain=PU.BL.IC.IP
domain=testers.com
allowexternaldomains=no
allowguest=no
avpf=yes
encryption=yes
transport=ws,udp
icesupport=yes
srvlookup=yes
And here's an example of a ws client in my realtime peer table:
id: 4
name: 660
ipaddr: PU.BL.IC.IP
port: 5060
regseconds: 1406368294
defaultuser: 660
fullcontact: sip:660 at PU.BL.IC.IP:5060
regserver:
useragent:
lastms: 0
host: dynamic
type: friend
context: default
deny: 0.0.0.0/0.0.0.0
permit: PU.BL.IC.IP
secret: NULL
md5secret: NULL
remotesecret: NULL
transport: NULL
dtmfmode: NULL
directmedia: NULL
nat: force_rport,comedia
callgroup: NULL
pickupgroup: NULL
language: NULL
disallow: NULL
allow: NULL
insecure: NULL
trustrpid: NULL
progressinband: NULL
promiscredir: NULL
useclientcode: NULL
accountcode: NULL
setvar: NULL
callerid: NULL
amaflags: NULL
callcounter: NULL
busylevel: NULL
allowoverlap: NULL
allowsubscribe: NULL
videosupport: NULL
maxcallbitrate: NULL
rfc2833compensate: NULL
mailbox: NULL
session-timers: NULL
session-expires: NULL
session-minse: NULL
session-refresher: NULL
t38pt_usertpsource: NULL
regexten: NULL
fromdomain: testers.com
fromuser: 660
qualify: NULL
defaultip: NULL
rtptimeout: NULL
rtpholdtimeout: NULL
sendrpid: NULL
outboundproxy: PU.BL.IC.IP
timert1: NULL
timerb: NULL
qualifyfreq: NULL
constantssrc: NULL
contactpermit: NULL
contactdeny: NULL
usereqphone: NULL
textsupport: NULL
faxdetect: NULL
buggymwi: NULL
auth: NULL
fullname: NULL
trunkname: NULL
cid_number: NULL
callingpres: NULL
mohinterpret: NULL
mohsuggest: NULL
parkinglot: NULL
hasvoicemail: NULL
subscribemwi: NULL
vmexten: NULL
autoframing: NULL
rtpkeepalive: NULL
call-limit: NULL
g726nonstandard: NULL
ignoresdpversion: NULL
allowtransfer: NULL
dynamic: NULL
path: NULL
supportpath: NULL
sippasswd: my-md5-pwd
rpid: NULL
domain: testers.com
sippasswd2: NULL
I'd greatly appreciate help on this!
cheers,
Olli
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20140726/27df226f/attachment.html>
Olli Heiskanen
2014-Aug-01 08:56 UTC
[asterisk-users] Rejecting secure audio stream without encryption details - when using ws clients and Kamailio integration
Hi, I got ahead with my setup, this post helped me much: http://forums.digium.com/viewtopic.php?f=1&t=90167&sid=66fdf8cc4be5d955ba584e989a23442f At least the avpf setting had to be removed from sip.conf and put in the realtime db table, defined per client. I left the encryption setting in sip.conf. I had some problems calling from SIP client to another, then had to define avpf=no for those clients. Personally I don't like to use different settings to different clients, is there a way around this? With this setup I can make calls between SIP clients but not ws clients. My client (now I use sip.js) fails to parse the sdp - including the apparently correct rtp profile UDP/TLS/RTP/SAVPF - and sends back 488, which makes the call fail. I'd like to hear opinions from you guys which would be the correct place to handle this? My setup has Asterisk Kamailio realtime integration, and I use dispatcher in Kamailio to route calls to Asterisk. Kamailio sounds like the logical place, but I'd rather find a way to not change the rtp profile along the way, at least until the clients can support that one. cheers, Olli 2014-07-26 12:58 GMT+03:00 Olli Heiskanen <ohjelmistoarkkitehti at gmail.com>:> > Greetings, > > I've noticed a problem that might originate from my Asterisk > configuration, could use a hand in sorting it out. Problem is a 488 > response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. > > My current setup has Asterisk Kamailio realtime integration, and Kamailio > uses dispatcher to route calls for Asterisk to handle. Now I have only one > Asterisk, on the same machine as Kamailio. The version is 11.10.2. With > Kamailio I use rtpengine, which affects SDP descriptions when 488 response > is received. > > My goal is to enable two websocket clients using Chrome to call each > other, using Kamailio as outbound proxy. Kamailio routes signaling to > Asterisk, and then back to clients. Currently the problem is RTP, when > INVITE is received from client A to Kamailio, it is relayed to Asterisk. > Asterisk responds with 488 Not Acceptable here and the cli says: > > NOTICE[11642][C-00000006]: chan_sip.c:10124 process_sdp: Received SAVPF > profle in audio offer but AVPF is not enabled, enabling: audio 30212 > RTP/SAVPF 111 103 104 0 8 106 105 13 126 > WARNING[11642][C-00000006]: chan_sip.c:10509 process_sdp: Rejecting > secure audio stream without encryption details: audio 30212 RTP/SAVPF 111 > 103 104 0 8 106 105 13 126 > > > Strange thing is, I don't know why Asterisk says AVPF is not enabled. The > warning about rejecting the audio stream must be behind the 488 response > but I didn't find any answers that would solve my case so I must turn to > you guys. In my sip.conf I have savpf=yes, but is there something else I > need to enable or change in the configs or change my peer configurations? > > I'm not sure if this is relevant but I checked that Asterisk was > successfully compiled with res_srtp module. > > Here's my sip.conf contents: > > bindport = 5070 ; using this since Kamailio is at 5060 > bindaddr = PU.BL.IC.IP > tcpenable = yes ;no > limitonpeers = yes > rtcachefriends = yes ; for realtime > rtupdate=yes > tos_sip=cs3 > tos_audio=ef > useragent=MyAsterisk > realm = myrealm.com > > autodomain=no > domain=PU.BL.IC.IP > domain=testers.com > > allowexternaldomains=no > allowguest=no > avpf=yes > encryption=yes > > transport=ws,udp > icesupport=yes > srvlookup=yes > > > And here's an example of a ws client in my realtime peer table: > > id: 4 > name: 660 > ipaddr: PU.BL.IC.IP > port: 5060 > regseconds: 1406368294 > defaultuser: 660 > fullcontact: sip:660 at PU.BL.IC.IP:5060 > regserver: > useragent: > lastms: 0 > host: dynamic > type: friend > context: default > deny: 0.0.0.0/0.0.0.0 > permit: PU.BL.IC.IP > secret: NULL > md5secret: NULL > remotesecret: NULL > transport: NULL > dtmfmode: NULL > directmedia: NULL > nat: force_rport,comedia > callgroup: NULL > pickupgroup: NULL > language: NULL > disallow: NULL > allow: NULL > insecure: NULL > trustrpid: NULL > progressinband: NULL > promiscredir: NULL > useclientcode: NULL > accountcode: NULL > setvar: NULL > callerid: NULL > amaflags: NULL > callcounter: NULL > busylevel: NULL > allowoverlap: NULL > allowsubscribe: NULL > videosupport: NULL > maxcallbitrate: NULL > rfc2833compensate: NULL > mailbox: NULL > session-timers: NULL > session-expires: NULL > session-minse: NULL > session-refresher: NULL > t38pt_usertpsource: NULL > regexten: NULL > fromdomain: testers.com > fromuser: 660 > qualify: NULL > defaultip: NULL > rtptimeout: NULL > rtpholdtimeout: NULL > sendrpid: NULL > outboundproxy: PU.BL.IC.IP > timert1: NULL > timerb: NULL > qualifyfreq: NULL > constantssrc: NULL > contactpermit: NULL > contactdeny: NULL > usereqphone: NULL > textsupport: NULL > faxdetect: NULL > buggymwi: NULL > auth: NULL > fullname: NULL > trunkname: NULL > cid_number: NULL > callingpres: NULL > mohinterpret: NULL > mohsuggest: NULL > parkinglot: NULL > hasvoicemail: NULL > subscribemwi: NULL > vmexten: NULL > autoframing: NULL > rtpkeepalive: NULL > call-limit: NULL > g726nonstandard: NULL > ignoresdpversion: NULL > allowtransfer: NULL > dynamic: NULL > path: NULL > supportpath: NULL > sippasswd: my-md5-pwd > rpid: NULL > domain: testers.com > sippasswd2: NULL > > > I'd greatly appreciate help on this! > > cheers, > Olli >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140801/e1309559/attachment.html>
Seemingly Similar Threads
- Letting rtp profiles be handled by rtpengine instead of Asterisk
- Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
- [webrtc] Received SAVPF profle in audio offer but AVPF is not enabled
- Asterisk removes ice lines in sdp when calling between webrtc clients
- From and To headers contain same account in INVITEs