tag:blogger.com,1999:blog-7262335442574749724.post4896177920025200786..comments2023-09-24T21:45:15.090+08:00Comments on SANJAY WILLIE'S Human Language. Asterisk | Nagios | OpenSource | Microsoft | Security: WebRTC and Asterisk 11 using sipML5 (with some FreePBX compatibility)JayWShttp://www.blogger.com/profile/04318296929423691109noreply@blogger.comBlogger85125tag:blogger.com,1999:blog-7262335442574749724.post-22163403905554154172014-07-12T19:12:11.083+08:002014-07-12T19:12:11.083+08:00Hello,
Have you managed to get this working with ...Hello,<br /><br />Have you managed to get this working with chrome? I tried all the settings you wrote, I have installed Asterisk 12 (I guess it doesn't require patching) with SRTP support, created certificatest for rtp, and in asterisk console I get message "Can't provide secure audio requested in SDP offer" and sipml5 says "Not acceptable here".<br /><br />Any workaround on this?<br /><br />Thanks!Mihailo Marinkovichttps://www.blogger.com/profile/03456152451596150171noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-83449448280288362222014-07-03T10:39:29.848+08:002014-07-03T10:39:29.848+08:00Hi ;
I try to call but i always have this prblm :...Hi ;<br /><br />I try to call but i always have this prblm :<br /><br />can't provide secure audio requested in sdp offer !? can u help me ? <br /><br />tnks <br />Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-41682370519542958912014-07-03T10:38:03.025+08:002014-07-03T10:38:03.025+08:00Hi ;
I try to call but i always have this prblm :...Hi ;<br /><br />I try to call but i always have this prblm :<br /><br />can't provide secure audio requested in sdp offer !? can u help me ? <br /><br />tnksAnonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-15869028368062580272014-06-16T21:30:53.072+08:002014-06-16T21:30:53.072+08:00Thanks for the tutorial.
Do you know is there any ...Thanks for the tutorial.<br><br />Do you know is there any progress on the warning:<br><br /><br /><b><br />chan_sip.c: Rejecting secure audio stream without encryption details: audio 59318 UDP/TLS/RTP/SAVPF 109 0 8 101<br /></b><br /><br />I am using wss with my own signed certificate.Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-10280167398987678582014-06-03T01:21:38.409+08:002014-06-03T01:21:38.409+08:00NOTE TO ALL, GOOGLE'S NEW UPDATE ON THEIR BROW...NOTE TO ALL, GOOGLE'S NEW UPDATE ON THEIR BROWSER REQUIRES DTLS A FORMAT WHICH IS DIFFERENT FROM THE PREVIOUS. I AM STILL TRYING TO FIGURE THIS ONE OUT. EVEN THE ASTERISK COMMUNITY ISN'T GIVING MUCH ATTENTION ON THIS ATM. SORRY.JayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-4589404537367337832014-05-28T10:15:05.294+08:002014-05-28T10:15:05.294+08:00WARNING[3756][C-0000000a]: chan_sip.c:10495 proces...WARNING[3756][C-0000000a]: chan_sip.c:10495 process_sdp: Rejecting secure audio stream without encryption details: audio 64393 RTP/SAVPF 111 103 104 0 ,,,,,,,,,,,,,,?????????????????Anonymoushttps://www.blogger.com/profile/15164159762099339931noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-68683996227695976982014-05-28T10:11:51.316+08:002014-05-28T10:11:51.316+08:00waaaaaaaaa3 my prbl is :
WARNING[3756][C-0000000a...waaaaaaaaa3 my prbl is :<br /><br />WARNING[3756][C-0000000a]: chan_sip.c:10495 process_sdp: Rejecting secure audio stream without encryption details: audio 64393 RTP/SAVPF 111 103 104 0 8 106 105 13 126<br /><br />whyyyyyyyyyyyyy ? tnks !Anonymoushttps://www.blogger.com/profile/15164159762099339931noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-9377442897459185742014-03-21T23:44:39.549+08:002014-03-21T23:44:39.549+08:00hi sanjay ...i install all setup according to your...hi sanjay ...i install all setup according to your given steps....but when i call from SIPml5 to ziper softphone the call is establish but not giving audio output...plz suggest me to troubleshoot this issue...thanxAnonymoushttps://www.blogger.com/profile/11093411002688242004noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-43854957015442295022014-03-03T19:43:37.632+08:002014-03-03T19:43:37.632+08:00Hi Sanjay,
Kindly suggest me to resolve the issue...Hi Sanjay,<br /><br />Kindly suggest me to resolve the issue,<br /><br />1. I used Ubuntu 12.04, Asterisk(asterisk-11.7.0) and SRTP.<br />2. Build successfully with srtp as your steps in local PC(192.168.2.221).<br />3. Created two users in sip.conf file.<br />4. Downloaded the sipml5 client sources and configured in same server(192.168.2.221).<br />5.Using sipml5 client at chrome browser register the users in two different PC browsers (Local Network) and It got registered too.<br />6. When try call from (8000) to another client (8008), the call is going and got a ringtone. but when I try allow from the other client then it got rejected.<br /><br />The follwoing error message am getting from the CLI mode,<br />[Mar 3 15:19:40] ERROR[8236][C-00000005]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known<br />[Mar 3 15:19:40] WARNING[8236][C-00000005]: chan_sip.c:15881 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'<br />[Mar 3 15:19:40] ERROR[8236][C-00000005]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supportedAnonymoushttps://www.blogger.com/profile/17415779781214372461noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-42099925494841349592014-03-03T19:41:55.872+08:002014-03-03T19:41:55.872+08:00Hi Sanjay,
Kindly suggest me to resolve the issue...Hi Sanjay,<br /><br />Kindly suggest me to resolve the issue,<br /><br />1. I used Ubuntu 12.04, Asterisk(asterisk-11.7.0) and SRTP.<br />2. Build successfully with srtp as your steps in local PC(192.168.2.221).<br />3. Created two users in sip.conf file.<br />4. Downloaded the sipml5 client sources and configured in same server(192.168.2.221).<br />5.Using sipml5 client at chrome browser register the users in two different PC browsers (Local Network) and It got registered too.<br />6. When try call from (8000) to another client (8008), the call is going and got a ringtone. but when I try allow from the other client then it got rejected.<br /><br />The follwoing error message am getting from the CLI mode,<br />[Mar 3 15:19:40] ERROR[8236][C-00000005]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known<br />[Mar 3 15:19:40] WARNING[8236][C-00000005]: chan_sip.c:15881 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'<br />[Mar 3 15:19:40] ERROR[8236][C-00000005]: netsock2.c:98 ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported<br /><br />Anonymoushttps://www.blogger.com/profile/17415779781214372461noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-86365796747313629212014-01-16T20:26:57.685+08:002014-01-16T20:26:57.685+08:00Hi There, first at all thank you very much to post...Hi There, first at all thank you very much to post this very nice way to implement WEBRtc. I would like to add something that happened to me in order to help to people who have this small failure. <br /><br />Sometimes at the end you have this problem: <br /><br />checking for the ability of -lsrtp to be linked in a shared object... no<br />configure: WARNING: ***<br />configure: WARNING: *** libsrtp could not be linked as a shared object.<br />configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp<br />configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr<br />configure: WARNING: *** replacing /usr with the prefix of your choice.<br />configure: WARNING: *** After re-installing libsrtp<br />configure: WARNING: *** configure script.<br />configure: WARNING: ***<br />configure: WARNING: *** If you do not need SRTP support re-run configure<br />configure: WARNING: *** with the --without-srtp option.<br /><br />To solve it, I did the following above before to use the command ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib<br /><br />In SRTP folder:<br /><br />make uninstall<br />make clean<br />./configure CFLAGS=-fPIC --prefix=/usr/local/lib<br />make<br />make runtest<br />make install<br /><br />After generating this change, using ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib will work without any problem. <br /><br />Cheers.Oswaldonoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-9668901151636407612014-01-13T19:30:27.342+08:002014-01-13T19:30:27.342+08:00Hi Sanjay, Thanks for this tutorial. I've been...Hi Sanjay, Thanks for this tutorial. I've been working at it for about 3 days now and your tutorial helped fix things along the way. So thank you very much for that.<br /><br />I have a question though. I am able to call successfully from the browser to a sip client (like Zoiper, Linphone etc). However, I am unable to make browser to browser calls or SIP phone to browser calls. At the moment I am using google chrome - Version 31.0.1650.63.<br /><br />Is there somewhere I may have gone wrong? Could you tell me what to look for? Asterisk's logs show nothing. This is what I see on my local chrome installation's logs:<br /><br />[6662:6699:0113/151954:ERROR:audio_manager_base.cc(417)] Not implemented reached in virtual std::string media::AudioManagerBase::GetAssociatedOutputDeviceID(const string&)<br />[6662:6691:0113/151954:ERROR:audio_manager_base.cc(422)] Not implemented reached in virtual std::string media::AudioManagerBase::GetDefaultOutputDeviceID()<br />[6662:6683:0113/151955:ERROR:audio_manager_base.cc(422)] Not implemented reached in virtual std::string media::AudioManagerBase::GetDefaultOutputDeviceID()<br />[6662:6683:0113/151955:ERROR:audio_manager_base.cc(422)] Not implemented reached in virtual std::string media::AudioManagerBase::GetDefaultOutputDeviceID()<br /><br />Anonymoushttps://www.blogger.com/profile/02775092262309327299noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-58558673174978024932013-10-18T23:26:09.163+08:002013-10-18T23:26:09.163+08:00Awesome! This worked like a charm for me!Awesome! This worked like a charm for me!Rafaelnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-33900311113347490652013-10-11T21:59:27.869+08:002013-10-11T21:59:27.869+08:00Can you please share the secret to get this workin...Can you please share the secret to get this working behind NAT? Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-91118396875409707152013-10-10T17:11:22.674+08:002013-10-10T17:11:22.674+08:00Thanks, I thought the Solution described here is a...Thanks, I thought the Solution described here is a complete solution where no additional hacks are required.Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-16365945594010665542013-10-10T00:24:25.077+08:002013-10-10T00:24:25.077+08:00haha I know how you feel, been having nightmares f...haha I know how you feel, been having nightmares figuring it out without hacking asterisk codes to get it work. Try googling it, James Mortensen had a good hack that worked for me... JayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-53598467155242821112013-10-07T17:00:56.746+08:002013-10-07T17:00:56.746+08:00Have you tried the sipml5 behind a NAT other than ...Have you tried the sipml5 behind a NAT other than your servers is in using its public IP? Or is all this just a theory?Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-64905839865409768422013-10-02T01:16:27.896+08:002013-10-02T01:16:27.896+08:00You dont need for LAN..think u need to.enable iceYou dont need for LAN..think u need to.enable iceJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-52775930022933723482013-10-02T00:54:26.257+08:002013-10-02T00:54:26.257+08:00You dont need for LAN..think u need to.enable iceYou dont need for LAN..think u need to.enable iceJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-38384563457981410842013-10-02T00:32:27.226+08:002013-10-02T00:32:27.226+08:00Hi,
in the rtp.conf file, the value of stunaddr, ...Hi,<br /><br />in the rtp.conf file, the value of stunaddr, do I really need a STUN server for a LAN setup ?<br /><br />If so, can I've a local STUN server, so it won't need any connection from outside ?Damjanhttps://www.blogger.com/profile/06340845175333672843noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-57776006105854810402013-10-01T20:46:12.012+08:002013-10-01T20:46:12.012+08:00Nice post. I have tried with different STUN but al...Nice post. I have tried with different STUN but all result same:<br /><br /><br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: chan_sip.c:11155 process_sdp_a_audio: Got Opus minptime=10<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 17081 RTP/SAVPF 111 103 104 0 8 106 105 13 126<br />Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-19967883274450817742013-10-01T20:44:49.222+08:002013-10-01T20:44:49.222+08:00Nice post.
I have tried with many STUN but result...Nice post. <br />I have tried with many STUN but result same:<br /><br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: chan_sip.c:11155 process_sdp_a_audio: Got Opus minptime=10<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 17081 RTP/SAVPF 111 103 104 0 8 106 105 13 126<br />Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-33476033899118884882013-10-01T20:43:51.276+08:002013-10-01T20:43:51.276+08:00Hi Sanjay
Nice post. I have tried and found
[Oc...Hi Sanjay<br /><br />Nice post. I have tried and found <br /><br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: sip/sdp_crypto.c:172 sdp_crypto_activate: Could not set SRTP policies<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: chan_sip.c:11155 process_sdp_a_audio: Got Opus minptime=10<br />[Oct 1 08:41:04] WARNING[28698][C-00000000]: chan_sip.c:10492 process_sdp: Rejecting secure audio stream without encryption details: audio 17081 RTP/SAVPF 111 103 104 0 8 106 105 13 126<br /><br /><br />Changed the stun server many but all result same.Anonymousnoreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-28207858708033429642013-09-09T02:00:54.115+08:002013-09-09T02:00:54.115+08:00Happy to report, this now works even on Android Ch...Happy to report, this now works even on Android Chorme (latest version). Audio seem distorted a little but we'll fix it, we'll fix it...{insane cool}JayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.comtag:blogger.com,1999:blog-7262335442574749724.post-20781310107748874592013-09-09T00:31:19.274+08:002013-09-09T00:31:19.274+08:00Jose,
Did u get SRTP working properly?
I just di...Jose,<br /><br />Did u get SRTP working properly?<br /><br />I just did the entire install using Asterisk 11.5 and I didn't have to patch, works straight away on latest Chrome 29.0.1547.66 m<br /><br />I will update the guide when I have some time. Very busy now :SJayWShttps://www.blogger.com/profile/04318296929423691109noreply@blogger.com