Help on freeswitch : gtalk-sip conversion
Hi, I am trying to get a call working from Gtalk client (Running on my laptop) to the SIP device(Registered as user name 1000). When gtalk client make a call, call is accepted by SIP device but the media is not flowing between Freeswitch and gtalk client. Please note that all the three (Freeswitch machine, gtalk client and SIP device) are in same private LAN inside the firewall. I see that freeswitch is rejecting the local candidate sent by gtalk client and selecting the stun candidate. Why is so ? How can I stop it from doing this. If it could select local candidate I feel call could establish. Thanks Sanjay
No reply on this but I moved with my debugging. Posting what i Found.... I found out that STUN (ICE) Binding requests coming from gtalk client were using RFC3489 but FS response was using RFC5389. In effect the response from FS were not quoting correct Transaction ID (First four bytes are MAGIC COOKIE) and hence gtalk not able to associate them. I corrected this in switch_stun.c and now I see once STUN response reach at gtalk client, it accepts them. I do not see call getting ended in 30 seconds and no more continuous ICE candidate Jabber messages from gtalk client! But user experience level I am still at same place - gtalk client still not sending its audio and RTP packets coming to it are not playing ! I think now its just media level issue. But what ? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
is selinux on freeswith node enabled ? it need to be disable On 9/28/12, sanjay <ssoni@lifesize.com> wrote:
No reply on this but I moved with my debugging. Posting what i Found....
I found out that STUN (ICE) Binding requests coming from gtalk client were using RFC3489 but FS response was using RFC5389. In effect the response from FS were not quoting correct Transaction ID (First four bytes are MAGIC COOKIE) and hence gtalk not able to associate them. I corrected this in switch_stun.c and now I see once STUN response reach at gtalk client, it accepts them. I do not see call getting ended in 30 seconds and no more continuous ICE candidate Jabber messages from gtalk client!
But user experience level I am still at same place - gtalk client still not sending its audio and RTP packets coming to it are not playing ! I think now its just media level issue. But what ?
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
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There is no selinux But apparmor is there and it is disabled. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
So, I could get audio working. I had to put in the client.xml. Actually, this forces crypto negotiation and audio flows both side. And I am able to have inbound (to FS) audio call from both gmail and gtalk. The additional modification needed for gtalk was - Change of code in FS to support RFC3489 (As I said in my above comment). Here I have all three nodes in my set-up (gmail, FS and SIP phone) in same LAN. This is the client.xml I used <profile type="client"> </profile> Now I want to try GV -> FS -> SIP . Can somebody help me what to change ? I have already configured GV to redirect calls to gmail chat and when I am logged-in in gmail chat, it gives a ring. However, GV is not redirecting calls to FS !! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
What did you put? sanjay wrote
So, I could get audio working. I had to put in the client.xml.
What????
Here I have all three nodes in my set-up (gmail, FS and SIP phone) in same LAN. This is the client.xml I used
<profile type="client">
</profile>
Now I want to try GV -> FS -> SIP . Can somebody help me what to change ? I have already configured GV to redirect calls to gmail chat and when I am logged-in in gmail chat, it gives a ring. However, GV is not redirecting calls to FS !!
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
Somehow, Everything which was under '<' and '>' is blank in my last post ! I put use_jingle=true in client.xml. I still need to try with when google client is not in same LAN. I think I need to play around with ACL for that, currently I have given rfc1918.auto. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
So, now just to close on this question. When gmail client / gtalk client is not in same LAN I should give param name="candidate-acl" value="wan.auto" param name="local-network-acl" value="localnet.auto" param name="ext-rtp-ip" value="$${external_rtp_ip}" Where external_rtp_ip is configured in vars.xml as stun:stun.freeswitch.org However, in some instances of the call there was no audio both side. ICE negotiation was not successful. But, some other time (When i Logout from both sides and login again), call was again with perfect audio. Need to check this instability! ------ But if gmail client is in same LAN then remove above three lines and provide param name="candidate-acl" value="rfc1918.auto" - With this set-up, calls always work perfectly :) NOTE: This is everything for inbound calls. I havent still tried out-bound calls. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-freeswitch-gtalk-sip-c... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (4)
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mazilo
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sanjay
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Sanjay Soni
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Tahir Almas