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- About Cristian Iconaru
08-18-2017
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Member since
05-28-2010
183
Posts
5
Helpful
0
Solutions
11-12-2015
03:51 AM
Hi Saravanan,
thanks for the gret info. I would have a question regarding the overhead on the controller processor when using unicast mode.
When would be actually optimal to switch to multicast mode? How many APs would be needed to reach that overhead? How many clients? Any raw guidelines?
Thanks.
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08-18-2014
02:16 AM
Hi, have you found any solution on this issue? I have it on SCCP42.9-2-1S also. Thanks.
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Created by
Cristian Iconaru
in VPN and AnyConnect
in VPN and AnyConnect
04-09-2014
03:55 AM
04-09-2014
03:55 AM
Hi all, is it possible to setup a site to site vpn in the following scenario? VPN-Router-1 ----- ASA (PAT) ----- VPN-Router-2 the ASA has a fixed IP and another IP range which is routed to it. would it be possible to build the vpn to the range that is routed to the ASA? it would be like this: ASA - 1.1.1.1 - static, configured on the outside interface. - 2.2.2.0/24 - range used for NATs/PAT, etc., routed to the ASA - would it work if the vpn will be forwarded from this range to the inside IP of VPN-Router-1? (static one-to-one NAT) Thanks. Regards
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Labels:
Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2014
03:45 AM
03-24-2014
03:45 AM
where exactly? I know this post but it's nothing related to this issue. I've tested the same config with asterisk and it works. Thanks.
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Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2014
01:35 AM
03-24-2014
01:35 AM
it doesn't really connect, it rings and when I answer the call gets terminated on the sip client side but the cisco phone stays "connected", as if it is connected. on webrtc console I got this, which is a bit strange because I only enbaled g711. so I believe something's wrong on webrtc side when I call from cisco -> sip. ***ERROR: function: "tdav_session_audio_start()" file: "src/audio/tdav_session_audio.c" line: "389" MSG: No codec matched ***ERROR: function: "tmedia_session_mgr_start()" file: "src/tmedia_session.c" line: "873" MSG: Failed to start audio session sip -> cisco, there's no message played, the call gets "declined" with SIP/2.0 401 Unauthorized. Thanks.
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Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-24-2014
12:55 AM
03-24-2014
12:55 AM
yes, it's the same as webrtc - 192.168.15.202. any ideas? Thanks.
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Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-19-2014
08:15 AM
03-19-2014
08:15 AM
Hi Manish, between sip client and webrtc gw -> ws and between webrtc gw and CM -> sip. here are the sip messages. both phones are registered, 9000 is a 7912 and 8080 is sip. 192.168.15.2 - CM 192.168.15.202 - webrtc SEND: INVITE sip:9000@192.168.15.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport From: <sip:8080@192.168.15.2>;tag=400660433 To: <sip:9000@192.168.15.2> Contact: <sip:8080@192.168.15.202:10060;ws-src-ip=192.168.251.105;ws-src-port=50731;ws-src-proto=ws;transport=udp> Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235 CSeq: 1875466830 INVITE Content-Type: application/sdp Content-Length: 978 Max-Forwards: 70 Authorization: Digest username="8080",realm="ccmsipline",nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6",uri="sip:9000@192.168.15.2",response="352cb2e17e36b32ee4e0d52443d0a106",algorithm=MD5 User-Agent: webrtc2sip Media Server 2.6.0 v=0 o=doubango 1983 678901 IN IP4 192.168.15.202 s=- c=IN IP4 192.168.15.202 t=0 0 a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF m=audio 58690 RTP/AVP 8 0 101 c=IN IP4 192.168.15.202 a=ptime:20 a=minptime:1 a=maxptime:255 a=silenceSupp:off - - - - a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-16 a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1YfBfgbhIdMB6YVtyZgJqc77QPHwm9o42aEPbkHD a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:fujGVOi70hQnKkeUimcFUw2bH3ajZ2iW0xKy5Nrw a=pcfg:1 t=1 a=1,2 a=pcfg:2 t=2 a=1,2 a=pcfg:3 t=3 a=sendrecv a=rtcp-mux a=ssrc:4034073057 cname:c08c56217e96dbc1e8234373eb5d2fcc a=ssrc:4034073057 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2 a=ssrc:4034073057 label:doubango@audio a=ice-ufrag:uaektHZ6KFVn1fw a=ice-pwd:HAj21nuOrDmIKl3ANXTc3K a=candidate:tWR5PLw1x 1 udp 2130706431 192.168.15.202 58690 typ host a=candidate:tWR5PLw1x 2 udp 2130706430 192.168.15.202 58691 typ host RECV:SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport From: <sip:8080@192.168.15.2>;tag=400660433 To: <sip:9000@192.168.15.2> Date: Wed, 19 Mar 2014 13:26:05 GMT Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235 CSeq: 1875466830 INVITE Allow-Events: presence Content-Length: 0 RECV:SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport From: <sip:8080@192.168.15.2>;tag=400660433 To: <sip:9000@192.168.15.2>;tag=856401750 Date: Wed, 19 Mar 2014 13:26:05 GMT Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235 CSeq: 1875466830 INVITE Allow-Events: presence WWW-Authenticate: Digest realm="ccmsipline", nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6", algorithm=MD5 Content-Length: 0 SEND: ACK sip:9000@192.168.15.2 SIP/2.0 Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport From: <sip:8080@192.168.15.2>;tag=400660433 To: <sip:9000@192.168.15.2>;tag=856401750 Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235 CSeq: 1875466830 ACK Content-Length: 0 Max-Forwards: 70 Receiving SIP o/ WebSocket message: ACK sip:9000@192.168.15.2 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKZEk81zTwfVde8oImts6ZHiTzchfBWh1N;rport From: "8080"<sip:8080@192.168.15.2>;tag=XFKqC4zu0S9QfzzMzQ4u To: <sip:9000@192.168.15.2>;tag=1464334432 Call-ID: ecc84fa2-3de3-d953-527f-5e7515cabca3 CSeq: 29519 ACK Content-Length: 0 Route: <sip:192.168.15.2:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 Thanks.
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Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-19-2014
07:19 AM
03-19-2014
07:19 AM
Third-party SIP Device. the phones do register on cm. (if the trunk is not set) Thanks
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Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-19-2014
07:08 AM
03-19-2014
07:08 AM
Hi Chris, thanks for the reply. well the devices are sipml5/html5 sip clients. I'm trying the webrtc with the CM in my lab to see how it works. The problem is that when I call from a sip client to a phone registered on the CM I've got: SIP/2.0 401 Unauthorized, so the call fails. I thought this might be due to the fact that the CM doesn't know the source IP of the INVITE. CM has no sip trunks. If I set up a trunk on the CM to the webrtc gateway, then the sip clients try to register over the trunk and then I've got: RECV:SIP/2.0 405 Method Not Allowed Warning: 399 cucm "SIP trunk disallows REGISTER" Any ideas? Do you have any experience with webrtc? Thanks.
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Created by
Cristian Iconaru
in IP Telephony and Phones
in IP Telephony and Phones
03-19-2014
05:38 AM
03-19-2014
05:38 AM
Hi guys, would it be possible to allow sip users to register over the sip trunk on the Call Manager? or is this method not allowed? Thanks. Best regards
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Labels:
Created by
Cristian Iconaru
in VPN and AnyConnect
in VPN and AnyConnect
11-20-2013
12:26 AM
11-20-2013
12:26 AM
thanks.
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Created by
Cristian Iconaru
in VPN and AnyConnect
in VPN and AnyConnect
11-19-2013
11:19 PM
11-19-2013
11:19 PM
Does anybody know this? Thanks.
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Created by
Cristian Iconaru
in VPN and AnyConnect
in VPN and AnyConnect
11-19-2013
11:27 AM
11-19-2013
11:27 AM
Hi Marius thanks for the reply, I mean the overall flash. for ex the ASA5500 needs 512MB: http://www.cisco.com/en/US/docs/security/vpn_client/anyconnect/anyconnect31/release/notes/anyconnect31rn.html#wp44010 The minimum flash memory recommended for all ASA 5500 models using AnyConnect 3.1 is 512MB but there's no info regarding the flash for routers. Does anybody know this? Thanks.
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11-12-2015
Hi Saravanan, thanks for the gret info. I would have a question
regarding the overhead on the contro...
Created by
Cristian Iconaru
in VPN and AnyConnect
04-09-2014
04-09-2014
Hi all,is it possible to setup a site to site vpn in the following
scenario?VPN-Router-1 ----- ASA (...
03-24-2014
where exactly? I know this post but it's nothing related to this
issue.I've tested the same config w...
03-24-2014
it doesn't really connect, it rings and when I answer the call gets
terminated on the sip client sid...
03-19-2014
Hi Manish,between sip client and webrtc gw -> ws and between webrtc gw
and CM -> sip.here are the si...
03-19-2014
Hi Chris,thanks for the reply.well the devices are sipml5/html5 sip
clients. I'm trying the webrtc w...
03-19-2014
Hi guys,would it be possible to allow sip users to register over the sip
trunk on the Call Manager? ...
Public Statistics
| Date Registered | 05-28-2010 08:28 AM |
| Date Last Visited | 08-18-2017 04:03 AM |
| Total Messages Posted | 183 |
| Total Helpful Votes Received | 5 |
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