As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration.
In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost):
REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;reg-id=1;+sip.instance="urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0
Asterisk answers this through UDP, and Kamailio forwards it through the websocket:
SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0
Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket:
NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws Contact: sip:asterisk@127.0.0.1:5080 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89
Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0)
Since I have not implemented handling of voicemail indications, the browser answers this:
SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws;tag=ggu5etber9 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0
After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws .
I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration.
Just in passing, why did you remove bindaddr=127.0.0.1?
In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost):
REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;reg-id=1;+sip.instance="urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0
Asterisk answers this through UDP, and Kamailio forwards it through the websocket:
SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0
Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket:
NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws Contact: sip:asterisk@127.0.0.1:5080 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89
Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0)
Since I have not implemented handling of voicemail indications, the browser answers this:
SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws;tag=ggu5etber9 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0
After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws .
I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?
What does the OPTIONS message in asterisk look like?
El 28/08/14 19:09, Paul Belanger escribió:
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration.
Just in passing, why did you remove bindaddr=127.0.0.1?
If I keep the bindaddr, then asterisk fails to send the DTLS-SRTP handshake packets, resulting in no audio. Apparently rtpproxy does not route this.
In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost):
REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;reg-id=1;+sip.instance="urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0
Asterisk answers this through UDP, and Kamailio forwards it through the websocket:
SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0
Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket:
NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws Contact: sip:asterisk@127.0.0.1:5080 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89
Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0)
Since I have not implemented handling of voicemail indications, the browser answers this:
SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws;tag=ggu5etber9 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0
After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws .
I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?
What does the OPTIONS message in asterisk look like?
elx3*CLI> sip qualify peer avillacisIM_pbx.villacis.com Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:68on862t@192.168.3.2:58927;transport=ws SIP/2.0 Via: SIP/2.0/WS 127.0.0.1:5080;branch=z9hG4bK2b267794;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as1a2c3be2 To: sip:68on862t@192.168.3.2:58927;transport=ws Contact: sip:asterisk@127.0.0.1:5080;transport=WS Call-ID: 7cbd63985b293b0150740e5a19143451@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.12.0 Date: Fri, 29 Aug 2014 15:54:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
---
On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
El 28/08/14 19:09, Paul Belanger escribió:
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration.
Just in passing, why did you remove bindaddr=127.0.0.1?
If I keep the bindaddr, then asterisk fails to send the DTLS-SRTP handshake packets, resulting in no audio. Apparently rtpproxy does not route this.
FWIW: I added a new setting into chan_sip, rptbindaddr[1], which allows you to no control the interface RTP binds too. Not sure if that helps in your setup or not.
In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost):
REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS
r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact:
sip:fnuql6ft@192.168.3.2:47307;transport=ws;reg-id=1;+sip.instance="urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0
Asterisk answers this through UDP, and Kamailio forwards it through the websocket:
SIP/2.0 200 OK Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS
r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0
Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket:
NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws Contact: sip:asterisk@127.0.0.1:5080 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89
Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0)
Since I have not implemented handling of voicemail indications, the browser answers this:
SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws;tag=ggu5etber9 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0
After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws .
I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?
What does the OPTIONS message in asterisk look like?
elx3*CLI> sip qualify peer avillacisIM_pbx.villacis.com Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:68on862t@192.168.3.2:58927;transport=ws SIP/2.0 Via: SIP/2.0/WS 127.0.0.1:5080;branch=z9hG4bK2b267794;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as1a2c3be2 To: sip:68on862t@192.168.3.2:58927;transport=ws Contact: sip:asterisk@127.0.0.1:5080;transport=WS Call-ID: 7cbd63985b293b0150740e5a19143451@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.12.0 Date: Fri, 29 Aug 2014 15:54:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
Ya, your via address is over the WS. What does your peer settings look like for avillacisIM_pbx.villacis.com ?
[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=422241
El 29/08/14 14:44, Paul Belanger escribió:
On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
El 28/08/14 19:09, Paul Belanger escribió:
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration.
Just in passing, why did you remove bindaddr=127.0.0.1?
If I keep the bindaddr, then asterisk fails to send the DTLS-SRTP handshake packets, resulting in no audio. Apparently rtpproxy does not route this.
FWIW: I added a new setting into chan_sip, rptbindaddr[1], which allows you to no control the interface RTP binds too. Not sure if that helps in your setup or not.
In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost):
REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS
r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact:
sip:fnuql6ft@192.168.3.2:47307;transport=ws;reg-id=1;+sip.instance="urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0
Asterisk answers this through UDP, and Kamailio forwards it through the websocket:
SIP/2.0 200 OK Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS
r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0
Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket:
NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws Contact: sip:asterisk@127.0.0.1:5080 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89
Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0)
Since I have not implemented handling of voicemail indications, the browser answers this:
SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws;tag=ggu5etber9 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0
After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws .
I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?
What does the OPTIONS message in asterisk look like?
elx3*CLI> sip qualify peer avillacisIM_pbx.villacis.com Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:68on862t@192.168.3.2:58927;transport=ws SIP/2.0 Via: SIP/2.0/WS 127.0.0.1:5080;branch=z9hG4bK2b267794;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as1a2c3be2 To: sip:68on862t@192.168.3.2:58927;transport=ws Contact: sip:asterisk@127.0.0.1:5080;transport=WS Call-ID: 7cbd63985b293b0150740e5a19143451@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.12.0 Date: Fri, 29 Aug 2014 15:54:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
Ya, your via address is over the WS. What does your peer settings look like for avillacisIM_pbx.villacis.com ?
[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=422241
mysql> select * from sip where name = 'avillacisIM_pbx.villacis.com'; +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ | id | name | context | callingpres | deny | permit | acl | secret | md5secret | remotesecret | transport | host | nat | type | accountcode | amaflags | callgroup | pickupgroup | namedcallgroup | namedpickupgroup | callerid | directmedia | directmediapermit | directmediaacl | description | defaultip | dtmfmode | fromuser | fromdomain | insecure | language | tonezone | mailbox | qualify | regexten | rtptimeout | rtpholdtimeout | setvar | disallow | allow | fullcontact | ipaddr | port | username | defaultuser | dial | trustrpid | sendrpid | progressinband | promiscredir | useclientcode | callcounter | busylevel | allowoverlap | allowsubscribe | allowtransfer | lastms | useragent | regseconds | regserver | videosupport | maxcallbitrate | rfc2833compensate | session-timers | session-expires | session-minse | session-refresher | outboundproxy | callbackextension | timert1 | timerb | qualifyfreq | constantssrc | contactpermit | contactdeny | contactacl | usereqphone | textsupport | faxdetect | buggymwi | auth | fullname | trunkname | cid_number | mohinterpret | mohsuggest | parkinglot | hasvoicemail | subscribemwi | vmexten | rtpkeepalive | g726nonstandard | ignoresdpversion | subscribecontext | template | keepalive | t38pt_usertpsource | organization_domain | outofcall_message_context | sippasswd | kamailioname | mwi_from | avpf | dtlsenable | dtlsverify | dtlscertfile | dtlsprivatekey | dtlssetup | force_avp | icesupport | encryption | +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ | 12 | avillacisIM_pbx.villacis.com | pbx.villacis.com-from-internal | NULL | 0.0.0.0/0.0.0.0 | 0.0.0.0/0.0.0.0 | NULL | NULL | NULL | NULL | ws,wss,udp | dynamic | force_rport,comedia | friend | NULL | NULL | NULL | NULL | NULL | NULL | device <avillacisIM> | no | NULL | NULL | NULL | NULL | auto | NULL | NULL | NULL | es | NULL | 101@pbx.villacis.com-default | no | NULL | 60 | 300 | NULL | all | ulaw,alaw,gsm | sip:uqcma3g6@192.168.3.2:59675^3Btransport=ws | 127.0.0.1 | 5060 | | avillacisIM | SIP/avillacisIM_pbx.villacis.com | yes | no | NULL | NULL | NULL | yes | NULL | no | NULL | yes | 0 | SIP.js/0.6.2 | 1409346610 | | yes | 384 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 60 | NULL | NULL | NULL | NULL | NULL | NULL | yes | NULL | NULL | 101 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | *97 | NULL | NULL | NULL | pbx.villacis.com-im-sip | NULL | NULL | NULL | pbx.villacis.com | pbx.villacis.com-im-sip | Avillacis12345 | avillacisIM | NULL | yes | yes | no | /etc/pki/tls/certs/localhost_asterisk.crt | /etc/pki/tls/private/localhost_asterisk.key | actpass | yes | yes | yes | +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ 1 row in set (0.00 sec)
[root@elx3 kamailio]# asterisk -rnx 'sip show peer avillacisIM_pbx.villacis.com'
* Name : avillacisIM_pbx.villacis.com Description : Realtime peer: Yes, cached Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pbx.villacis.com-from-internal Record On feature : automon Record Off feature : automon Subscr.Cont. : pbx.villacis.com-im-sip Language : es Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : 101@pbx.villacis.com-default VM Extension : *97 LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "101" <avillacisIM> MaxCallBR : 384 kbps Expire : 153 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : No TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : auto Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 127.0.0.1:5060 Defaddr->IP : (null) Prim.Transp. : WS Allowed.Trsp : UDP,WS,WSS Def. Username: avillacisIM SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent : SIP.js/0.6.2 Reg. Contact : sip:uqcma3g6@192.168.3.2:59675;transport=ws Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : Yes Ign.Lifetime : No
I think the situation is because of the change of transport. How should this be handled so that Asterisk stops trying to use websocket transport for the signaling that came from the UDP port?
On Fri, Aug 29, 2014 at 5:08 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
El 29/08/14 14:44, Paul Belanger escribió:
On Fri, Aug 29, 2014 at 11:55 AM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
El 28/08/14 19:09, Paul Belanger escribió:
On Thu, Aug 28, 2014 at 7:18 PM, Alex Villacís Lasso a_villacis@palosanto.com wrote:
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed the bindaddr=127.0.0.1 from sip.conf and instead put a deny=0.0.0.0/0.0.0.0 and permit=127.0.0.1/255.255.255.0 in order to restrict SIP signaling to localhost. This allows asterisk to bypass rtpproxy when signaling through a websocket. I have already established calls originating from the browser. However, I have an issue with the registration.
Just in passing, why did you remove bindaddr=127.0.0.1?
If I keep the bindaddr, then asterisk fails to send the DTLS-SRTP handshake packets, resulting in no audio. Apparently rtpproxy does not route this.
FWIW: I added a new setting into chan_sip, rptbindaddr[1], which allows you to no control the interface RTP binds too. Not sure if that helps in your setup or not.
In my setup, Kamailio receives the REGISTER from whatever source, and forwards this through UDP to Asterisk, after the multiple-domain transformation. Therefore, Asterisk sees the following in its SIP port (all traffic through localhost):
REGISTER sip:pbx.villacis.com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0 Via: SIP/2.0/WSS
r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 Max-Forwards: 69 To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Contact:
sip:fnuql6ft@192.168.3.2:47307;transport=ws;reg-id=1;+sip.instance="urn:uuid:6b0c58ee-bdc5-47c0-aff0-963132dc0cad";expires=600 Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE,MESSAGE Supported: path,gruu,outbound User-Agent: SIP.js/0.6.2 Content-Length: 0
Asterisk answers this through UDP, and Kamailio forwards it through the websocket:
SIP/2.0 200 OK Via: SIP/2.0/UDP
127.0.0.1;branch=z9hG4bKc1c5.cb49f656197d0ba16f2a1661dd6a44cc.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/WSS
r01r0mla9hdp.invalid;rport=47307;received=192.168.3.2;branch=z9hG4bK9309681 From: "Alex Villac..s" sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=b5c0lq4kac To: sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080;tag=as5ae2df76 Call-ID: vp2akar0aqfmgfa6m1taau CSeq: 82 REGISTER Server: Asterisk PBX 11.12.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 600 Contact: sip:fnuql6ft@192.168.3.2:47307;transport=ws;expires=600 Date: Thu, 28 Aug 2014 22:21:15 GMT Content-Length: 0
Then Asterisk sends this through UDP, and Kamailio again forwards it through the websocket:
NOTIFY sip:fnuql6ft@192.168.3.2:47307;transport=ws SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws Contact: sip:asterisk@127.0.0.1:5080 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.12.0 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89
Messages-Waiting: no Message-Account: sip:*97@127.0.0.1:5080 Voice-Message: 0/0 (0/0)
Since I have not implemented handling of voicemail indications, the browser answers this:
SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK4d60f167;rport=5080 To: sip:fnuql6ft@192.168.3.2:47307;transport=ws;tag=ggu5etber9 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as43c12840 Call-ID: 04deeb0068a847fa514d748c7d9993c5@127.0.0.1:5080 CSeq: 102 NOTIFY Supported: outbound Content-Length: 0
After that, Asterisk wants to send an OPTIONS packet. From the point of view of Asterisk (sip set debug on), it is already sent, but never gets a response. However, tcpdump shows that the packet is never sent through the localhost interface in the first place. It is also not sent through any other interface. My guess is that since the REGISTER has a contact with transport=ws , Asterisk wants to send this through a websocket (which is disabled). So I could have to generate a contact without transport=ws .
I have worked around this by setting qualify=no in the account for the websocket, but I would like a better solution, one that allows the OPTIONS packet to reach the browser, and to get the response. What is the proper way to deal with this?
What does the OPTIONS message in asterisk look like?
elx3*CLI> sip qualify peer avillacisIM_pbx.villacis.com Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:68on862t@192.168.3.2:58927;transport=ws SIP/2.0 Via: SIP/2.0/WS 127.0.0.1:5080;branch=z9hG4bK2b267794;rport Max-Forwards: 70 From: "asterisk" sip:asterisk@127.0.0.1:5080;tag=as1a2c3be2 To: sip:68on862t@192.168.3.2:58927;transport=ws Contact: sip:asterisk@127.0.0.1:5080;transport=WS Call-ID: 7cbd63985b293b0150740e5a19143451@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.12.0 Date: Fri, 29 Aug 2014 15:54:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0
Ya, your via address is over the WS. What does your peer settings look like for avillacisIM_pbx.villacis.com ?
[1] http://svnview.digium.com/svn/asterisk?view=revision&revision=422241
mysql> select * from sip where name = 'avillacisIM_pbx.villacis.com'; +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ | id | name | context | callingpres | deny | permit | acl | secret | md5secret | remotesecret | transport | host | nat | type | accountcode | amaflags | callgroup | pickupgroup | namedcallgroup | namedpickupgroup | callerid | directmedia | directmediapermit | directmediaacl | description | defaultip | dtmfmode | fromuser | fromdomain | insecure | language | tonezone | mailbox | qualify | regexten | rtptimeout | rtpholdtimeout | setvar | disallow | allow | fullcontact | ipaddr | port | username | defaultuser | dial | trustrpid | sendrpid | progressinband | promiscredir | useclientcode | callcounter | busylevel | allowoverlap | allowsubscribe | allowtransfer | lastms | useragent | regseconds | regserver | videosupport | maxcallbitrate | rfc2833compensate | session-timers | session-expires | session-minse | session-refresher | outboundproxy | callbackextension | timert1 | timerb | qualifyfreq | constantssrc | contactpermit | contactdeny | contactacl | usereqphone | textsupport | faxdetect | buggymwi | auth | fullname | trunkname | cid_number | mohinterpret | mohsuggest | parkinglot | hasvoicemail | subscribemwi | vmexten | rtpkeepalive | g726nonstandard | ignoresdpversion | subscribecontext | template | keepalive | t38pt_usertpsource | organization_domain | outofcall_message_context | sippasswd | kamailioname | mwi_from | avpf | dtlsenable | dtlsverify | dtlscertfile | dtlsprivatekey | dtlssetup | force_avp | icesupport | encryption | +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ | 12 | avillacisIM_pbx.villacis.com | pbx.villacis.com-from-internal | NULL | 0.0.0.0/0.0.0.0 | 0.0.0.0/0.0.0.0 | NULL | NULL | NULL | NULL | ws,wss,udp | dynamic | force_rport,comedia | friend | NULL | NULL
This is your issue, you are allowing ws and wss as the transport. Because you are forwarding calls over UDP, that is the only protocol you should be allowing.
| NULL | NULL | NULL | NULL | device <avillacisIM> | no | NULL | NULL | NULL | NULL | auto | NULL | NULL | NULL | es | NULL | 101@pbx.villacis.com-default | no | NULL | 60 | 300 | NULL | all | ulaw,alaw,gsm | sip:uqcma3g6@192.168.3.2:59675^3Btransport=ws | 127.0.0.1 | 5060 | | avillacisIM | SIP/avillacisIM_pbx.villacis.com | yes | no | NULL | NULL | NULL | yes | NULL | no | NULL | yes | 0 | SIP.js/0.6.2 | 1409346610 | | yes | 384 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 60 | NULL | NULL | NULL | NULL | NULL | NULL | yes | NULL | NULL | 101 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | *97 | NULL | NULL | NULL | pbx.villacis.com-im-sip | NULL | NULL | NULL | pbx.villacis.com | pbx.villacis.com-im-sip | Avillacis12345 | avillacisIM | NULL | yes | yes | no | /etc/pki/tls/certs/localhost_asterisk.crt | /etc/pki/tls/private/localhost_asterisk.key | actpass | yes | yes | yes | +----+------------------------------+--------------------------------+-------------+-----------------+-----------------+------+--------+-----------+--------------+------------+---------+---------------------+--------+-------------+----------+-----------+-------------+----------------+------------------+----------------------+-------------+-------------------+----------------+-------------+-----------+----------+----------+------------+----------+----------+----------+------------------------------+---------+----------+------------+----------------+--------+----------+---------------+-----------------------------------------------+-----------+------+----------+-------------+----------------------------------+-----------+----------+----------------+--------------+---------------+-------------+-----------+--------------+----------------+---------------+--------+--------------+------------+-----------+--------------+----------------+-------------------+----------------+-----------------+---------------+-------------------+---------------+-------------------+---------+--------+-------------+--------------+---------------+-------------+------------+-------------+-------------+-----------+----------+------+----------+-----------+------------+--------------+------------+------------+--------------+--------------+---------+--------------+-----------------+------------------+-------------------------+----------+-----------+--------------------+---------------------+---------------------------+----------------+--------------+----------+------+------------+------------+-------------------------------------------+---------------------------------------------+-----------+-----------+------------+------------+ 1 row in set (0.00 sec)
[root@elx3 kamailio]# asterisk -rnx 'sip show peer avillacisIM_pbx.villacis.com'
- Name : avillacisIM_pbx.villacis.com
Description : Realtime peer: Yes, cached Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pbx.villacis.com-from-internal Record On feature : automon Record Off feature : automon Subscr.Cont. : pbx.villacis.com-im-sip Language : es Tonezone : <Not set> AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Named Callgr : Nam. Pickupgr: MOH Suggest : Mailbox : 101@pbx.villacis.com-default VM Extension : *97 LastMsgsSent : 0/0 Call limit : 2147483647 Max forwards : 0 Dynamic : Yes Callerid : "101" <avillacisIM> MaxCallBR : 384 kbps Expire : 153 Insecure : no Force rport : Yes Symmetric RTP: Yes ACL : Yes DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 4294967295 DirectMedia : No PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : Yes Send RPID : No TrustIDOutbnd: Legacy Subscriptions: Yes Overlap dial : No DTMFmode : auto Timer T1 : 500 Timer B : 32000 ToHost : Addr->IP : 127.0.0.1:5060 Defaddr->IP : (null) Prim.Transp. : WS Allowed.Trsp : UDP,WS,WSS
Again, see above, this should only be udp.
Def. Username: avillacisIM SIP Options : (none) Codecs : (gsm|ulaw|alaw) Codec Order : (ulaw:20,alaw:20,gsm:20) Auto-Framing : No Status : Unmonitored Useragent : SIP.js/0.6.2 Reg. Contact : sip:uqcma3g6@192.168.3.2:59675;transport=ws Qualify Freq : 60000 ms Keepalive : 0 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : Yes Ign.Lifetime : No
I think the situation is because of the change of transport. How should this be handled so that Asterisk stops trying to use websocket transport for the signaling that came from the UDP port?
See my responses above.
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Alex Villacís Lasso -
Paul Belanger