Help with video call ( h.264 codec )
Hi guys, i am configuring a video call between two grandstream video phones (GXV-3000) which i don't recommend for being a crappy set of devices. I am using the latest freeswitch FreeSWITCH version 1.2.0rc2+git~20120618T111915Z~11a745cf76+unclean~20120618T121534Z . But no video call between those devices. However video call between a Grandstream and an Android Bria Phone works ( i don't recomment bria too, but it helps me with some tests). Here is the log of two grandstream that fails: http://pastebin.com/tXZ4DhKM [1] it keeps stuck on the line that says: 2012-06-18 11:58:35.144424 [DEBUG] sofia.c:5838 Channel sofia/internal/215 entering state [calling][0] and then continues. And here is the log of a grandstream and a bria that works: http://pastebin.com/nRzSr7WR [2] The call is started from the Bria, starting the call from Grandstream will not work. If i configure the Grandstream to not send video they can place calls and it will work for audio, if i configure them to start video automatically they will not work, in fact they can't place any call but they can receive calls from other phones. But if i put: <param name="inbound-bypass-media" value="true"/> at the sip profile they are attached to they will work, however i don't like this solution, cause i want to deploy them behind a firewall at different places. Tried before: <param name="inbound-proxy-media" value="true"/> but it didn't work, the only way to have this working is setting inbound_bypass_media. Any help on this issue ? Thanks. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57.1-485-3020 x 211 Cel. +57.316-227-3595 nestor@tiendalinux.com www.tiendalinux.com Links: ------ [1] http://pastebin.com/tXZ4DhKM [2] http://pastebin.com/nRzSr7WR
Just to add the freeswitch keep sending udp messages to this phone, but the phone never process those messages, i have configured the phone to log at the debug level and it logs every sip message, but none of the messages sent from freeswitch, i use tcpdump and confirmed the packages went out from FS and sent to the phone, but the phone never processed those messages ... well anyway those are grandstream phones, a total piece of junk. This is the message FS keep sending . send 1490 bytes to udp/[172.24.103.44]:5060 at 22:30:23.032784: ------------------------------------------------------------------------ INVITE sip:215@172.24.103.44:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.24.107.1:5090;rport;branch=z9hG4bK8tZND2F7QccBQ Max-Forwards: 69 From: "Ext. 216" <sip:216@172.24.107.1>;tag=cyQ3Fagm3FBvr To: <sip:215@172.24.103.44:5060;transport=udp> Call-ID: fff8c62a-3437-1230-1388-00224d56ac0c CSeq: 29693240 INVITE Contact: <sip:mod_sofia@172.24.107.1:5090> User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120618T111915Z~11a745cf76+unclean~20120618T121534Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 511 X-FS-Support: update_display,send_info Remote-Party-ID: "Ext. 216" <sip:216@172.24.107.1>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1569664329 1569664330 IN IP4 172.24.107.1 s=FreeSWITCH c=IN IP4 172.24.107.1 t=0 0 m=audio 12058 RTP/AVP 0 8 4 18 3 2 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=ptime:20 m=video 12044 RTP/AVP 99 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg== a=framerate:15 ------------------------------------------------------------------------ -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57.1-485-3020 x 211 Cel. +57.316-227-3595 nestor@tiendalinux.com www.tiendalinux.com
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Nestor A. Diaz