10 Feb
2014
10 Feb
'14
2 p.m.
I am having problems with calls from webrtc to kamailio forwarded to Asterisk
These are snippet of the debug logs
Asterisk
CSeq: 4910 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
Jssip
Cause: Bad Media Description
Origin: remote
Searching on google I get some indication this is to do with ice config?
Please can some one suggest if this is so.
In my scenerio the webrt clients will only call to the asterisk server (and not to other
user agent).
Considersing this I think maybe can do without ice.
Is it possbile to disable ice.
11 Feb
11 Feb
11:48 p.m.
I'm familiar with Freeswitch, not Asterisk. So, I don't know my comment
will be applicable there.
But, could you explain your signaling path a little. Is websocket being
handled by Asterisk or somebody else in between. In my case, there is
Kamailio in between FS and webRTC client. So, Freeswitch was modifying the
SDP to non-webRTC, so called webRTC client rejected the call. I had to set
FS to media proxy mode to stop it from modifying SDP.
Thanks,
Dipak
On Mon, Feb 10, 2014 at 8:00 AM, jaflong jaflong <jaflong(a)yandex.com> wrote:
I am having problems with calls from webrtc to
kamailio forwarded to
Asterisk
These are snippet of the debug logs
Asterisk
CSeq: 4910 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
Jssip
Cause: Bad Media Description
Origin: remote
Searching on google I get some indication this is to do with ice config?
Please can some one suggest if this is so.
In my scenerio the webrt clients will only call to the asterisk server
(and not to other user agent).
Considersing this I think maybe can do without ice.
Is it possbile to disable ice.
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Thanks,
Dipak
12 Feb
12 Feb
12:38 p.m.
I guess ice is implemented at web browser level, not sure there is an
option to control it from java script.
On the other hand, you should run asterisk in debug mode and see what it
is printing. It doesn't seem an issue related to kamailio at all.
Cheers,
Daniel
On 10/02/14 14:00, jaflong jaflong wrote:
I am having problems with calls from webrtc to
kamailio forwarded to Asterisk
These are snippet of the debug logs
Asterisk
CSeq: 4910 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.0
Content-Length: 0
Jssip
Cause: Bad Media Description
Origin: remote
Searching on google I get some indication this is to do with ice config?
Please can some one suggest if this is so.
In my scenerio the webrt clients will only call to the asterisk server (and not to other
user agent).
Considersing this I think maybe can do without ice.
Is it possbile to disable ice.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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2 comments
3 participants
participants (3)
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Biswas, Dipak -
Daniel-Constantin Mierla -
jaflong jaflong