Problems with proxy media and bypass media in FreeSWITCH
I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/dialplan/default.xml: <extension name="setup_media" continue="true"> <condition field="${sip_nat_detected}" expression="true"> <action application="set" data="proxy_media=true" /> <action application="set" data="bypass_media=false" /> <anti-action application="set" data="proxy_media=false" /> <anti-action application="set" data="bypass_media=true" /> </condition> </extension> I have the following configured in /usr/local/freeswitch/conf/vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,iLBC@20i,G722,PCMU,PCMA"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,iLBC@20i,G722,PCMU,PCMA"/> Here is the SIP trace for the failing call: Nov 23 17:55:05.245 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:+19725357722@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24 Record-Route: <sip:65.211.120.237:5060;lr> v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236 record-route: <sip:63.77.76.236;lr> f: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a t: <sip:+19725357722@63.77.76.236:5060;user=phone> i: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Max-Forwards: 16 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 m: <sip:199.173.101.208:5060;transport=UDP> c: application/SDP l: 210 P-Asserted-Identity: <sip:9729831777@63.77.76.236;user=phone> Privacy: none v=0 o=- 641026559 641026559 IN IP4 199.173.111.147 s=- c=IN IP4 199.173.111.147 t=0 0 m=audio 33344 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 From: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a To: <sip:+19725357722@63.77.76.236:5060;user=phone> Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 17:55:05.257 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212> Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 Supported: timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 1961129755-3619819998-2727664095-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259020505 Contact: <sip:19729831777@168.75.202.246:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 15 P-Asserted-Identity: <sip:19729831777@168.75.202.246> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 5041 5861 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.147 t=0 0 m=audio 33344 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.147 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:05.261 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212> Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 CSeq: 101 INVITE Timestamp: 1259020505 0.000345 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212>;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 CSeq: 101 INVITE Contact: <sip:19725357722@168.75.202.212:5062;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 P-Asserted-Identity: "19725357722" <sip:19725357722@168.75.202.212> Nov 23 17:55:05.309 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 65.211.120.237:5060;branch=z9hG4bKec920f9119165c414d2f6229bb6a76ac.8e1ce24,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK19a30c0f46372620ff158f019d0ce5df.24ee0396;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3d9f0a-f0b542be-62e0db38;received=199.173.101.208 From: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3d9f0a-5460a3be-3d9f0a To: <sip:+19725357722@63.77.76.236:5060;user=phone>;tag=105BD180-BD7 Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: a14d9878d065adc713c43d9f0af0b542beb67295e9c2c7438-0569-6585 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:19725357722@168.75.202.246:5060> Record-Route: <sip:65.211.120.237:5060;lr>,<sip:63.77.76.236;lr> Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659A1F3 From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212>;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 CSeq: 101 INVITE Contact: <sip:19725357722@168.75.202.212:5062;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 202 P-Asserted-Identity: "19725357722" <sip:19725357722@168.75.202.212> v=0 o=- 211627 211627 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729a/8000 a=fmtp:96 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722@168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659B6A From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212>;tag=DFKSy9Q5DK1Na Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 Nov 23 17:55:08.397 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:19725357722@168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212>;tag=DFKSy9Q5DK1Na Date: Mon, 23 Nov 2009 23:55:05 GMT Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 P-Asserted-Identity: <sip:19729831777@168.75.202.246> Timestamp: 1259020508 CSeq: 102 BYE Reason: Q.850;cause=65 Content-Length: 0 Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212>;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 CSeq: 102 BYE Timestamp: 1259020508 0.000093 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 17:55:08.401 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK659C1B3D From: <sip:19729831777@168.75.202.246>;tag=105BD148-201C To: <sip:19725357722@168.75.202.212>;tag=DFKSy9Q5DK1Na Call-ID: 74E5B003-D7C211DE-A29AD9DF-3419A306@168.75.202.246 CSeq: 102 BYE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 The SIP call actually fails. If I remove the following from /usr/local/freeswitch/conf/dialplan/default.xml: <extension name="setup_media" continue="true"> <condition field="${sip_nat_detected}" expression="true"> <action application="set" data="proxy_media=true" /> <action application="set" data="bypass_media=false" /> <anti-action application="set" data="proxy_media=false" /> <anti-action application="set" data="bypass_media=true" /> </condition> </extension> And I change this line in /usr/local/freeswitch/conf/vars.xml from <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,iLBC@20i,G722,PCMU,PCMA"/> to <X-PRE-PROCESS cmd="set" data="global_codec_prefs=iLBC@20i,G722,PCMU,PCMA"/> And I change this line in /usr/local/freeswitch/conf/vars.xml from <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,iLBC@20i,G722,PCMU,PCMA"/> to <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=iLBC@20i,G722,PCMU,PCMA"/> both inbound and outbound calls succeed. Here is a SIP trace of a successful call after I apply the above changes: Nov 23 18:16:51.844 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: INVITE sip:+19725357722@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb Record-Route: <sip:65.243.172.245:5060;lr> v: SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205 record-route: <sip:65.217.40.205;lr> f: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 t: <sip:+19725357722@65.217.40.205:5060;user=phone> i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Max-Forwards: 16 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 m: <sip:199.173.101.208:5060;transport=UDP> c: application/SDP l: 210 P-Asserted-Identity: <sip:9729831777@65.217.40.205;user=phone> Privacy: none v=0 o=- 654094598 654094598 IN IP4 199.173.111.138 s=- c=IN IP4 199.173.111.138 t=0 0 m=audio 31456 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 18:16:51.852 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 To: <sip:+19725357722@65.217.40.205:5060;user=phone> Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357722@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 To: <sip:19725357722@168.75.202.212> Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 Supported: timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 2142226702-3620016606-2730940895-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259021811 Contact: <sip:19729831777@168.75.202.246:5060> Expires: 180 Allow-Events: telephone-event Max-Forwards: 15 P-Asserted-Identity: <sip:19729831777@168.75.202.246> Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 9668 3852 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.138 t=0 0 m=audio 31456 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.138 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 23 18:16:51.856 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 To: <sip:19725357722@168.75.202.212> Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 CSeq: 101 INVITE Timestamp: 1259021811 0.000356 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Content-Length: 0 Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 To: <sip:19725357722@168.75.202.212>;tag=mXNXUN859rKBa Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 CSeq: 101 INVITE Contact: <sip:19725357722@168.75.202.212:5062;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Content-Length: 0 P-Asserted-Identity: "19725357722" <sip:19725357722@168.75.202.212> Nov 23 18:16:51.908 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 To: <sip:+19725357722@65.217.40.205:5060;user=phone>;tag=106FC168-50D Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:19725357722@168.75.202.246:5060> Record-Route: <sip:65.243.172.245:5060;lr>,<sip:65.217.40.205;lr> Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 23 18:16:54.408 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AD9D0 From: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 To: <sip:19725357722@168.75.202.212>;tag=mXNXUN859rKBa Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 CSeq: 101 INVITE Contact: <sip:19725357722@168.75.202.212:5062;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 251 P-Asserted-Identity: "19725357722" <sip:19725357722@168.75.202.212> v=0 o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212 s=FreeSWITCH c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357722@168.75.202.212:5062;transport=udp SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AE109A From: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 To: <sip:19725357722@168.75.202.212>;tag=mXNXUN859rKBa Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 Nov 23 18:16:54.412 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bK1d6dd953ef66469db06038ec3bd2ec49.6fb41dbb,SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK74e3354e4bab8f6d8afec83d314c15b8.6f41c9d4;received=65.217.40.205,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da425-f0c931a9-52a4c353;received=199.173.101.208 From: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 To: <sip:+19725357722@65.217.40.205:5060;user=phone>;tag=106FC168-50D Date: Tue, 24 Nov 2009 00:16:51 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:19725357722@168.75.202.246:5060> Record-Route: <sip:65.243.172.245:5060;lr>,<sip:65.217.40.205;lr> Supported: replaces Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 253 v=0 o=CiscoSystemsSIP-GW-UserAgent 7353 3710 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 c=IN IP4 168.75.202.212 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Nov 23 18:16:54.492 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: ACK sip:19725357722@168.75.202.246:5060 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKb17fac77c446113b9154e16639d30287.6be1820d v: SIP/2.0/UDP 65.217.40.205:5060;branch=z9hG4bK99af095cadf31f4291c6a809ef6a6e03.7c44d9fc;received=65.217.40.205 f: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 t: <sip:+19725357722@65.217.40.205:5060;user=phone>;tag=106FC168-50D i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 1 ACK user-agent: CS2000_NGSS/9.0 Max-Forwards: 68 k: 100rel,replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3da428-f0c93cc6-61239372;received=199.173.101.208 m: <sip:199.173.101.208:5060;transport=UDP> l: 0 Nov 23 18:17:00.636 CST: %FAN-3-FAN_FAILED: Fan 1 had a rotation error reported. Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: BYE sip:19729831777@168.75.202.246:5060 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ Max-Forwards: 70 From: <sip:19725357722@168.75.202.212>;tag=mXNXUN859rKBa To: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 CSeq: 123392377 BYE Contact: <sip:19725357722@168.75.202.212:5062;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15586M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Nov 23 18:17:06.748 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 200 OK Via: SIP/2.0/UDP 168.75.202.212:5062;rport;branch=z9hG4bKmNaaUQK5vrpXQ From: <sip:19725357722@168.75.202.212>;tag=mXNXUN859rKBa To: <sip:19729831777@168.75.202.246>;tag=106FC130-1578 Date: Tue, 24 Nov 2009 00:17:06 GMT Call-ID: 7FB1015E-D7C511DE-A2CCD9DF-3419A306@168.75.202.246 Server: Cisco-SIPGateway/IOS-12.x CSeq: 123392377 BYE Reason: Q.850;cause=16 Content-Length: 0 Nov 23 18:17:06.752 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Sent: BYE sip:199.173.101.208:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK65AF4FA From: <sip:+19725357722@65.217.40.205:5060;user=phone>;tag=106FC168-50D To: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 Date: Tue, 24 Nov 2009 00:16:54 GMT Call-ID: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Route: <sip:65.243.172.245:5060;lr>,<sip:65.217.40.205;lr> Timestamp: 1259021826 CSeq: 101 BYE Reason: Q.850;cause=16 Content-Length: 0 Nov 23 18:17:06.824 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 200 OK f: <sip:+19725357722@65.217.40.205:5060;user=phone>;tag=106FC168-50D t: <sip:+19729831777@199.173.101.208:5060;user=phone>;tag=dc7-13c4-3da425-183eff65-3da425 i: 9fffb808d065adc713c43da425f0c931a91220c864c201e88-0568-4457 CSeq: 101 BYE server: CS2000_NGSS/9.0 allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP 168.75.202.246:5060;received=168.75.202.246;branch=z9hG4bK65AF4FA l: 0 I compared the SIP messaging from the failed call to the SIP messaging from the good call. Both calls are inbound calls. Here is the session description for the failed inbound call: v=0 o=- 211627 211627 IN IP4 192.168.1.4 s=- c=IN IP4 173.57.44.212 t=0 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 G729a/8000 a=fmtp:96 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Here is the session description for the good outbound call: v=0 o=FreeSWITCH 1259003870 1259003871 IN IP4 168.75.202.212 s=FreeSWITCH c=IN IP4 168.75.202.212 t=0 0 m=audio 17544 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Here are the differences between the session descriptors of the failed call and the good call: - The c= line has the correct IP address for the failed call, which was using media bypass - The c= line has the correct IP address for the good call, because the media is being processed by FreeSWITCH in the good call - The m= line does not have the correct RTP port in the failed call - The m= line has the correct RTP port in the good call I noticed that the SDP media descriptor is incorrect in the failed call. Has this problem been fixed? I am running revision 15586 from the FreeSWITCH SVN trunk. _________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/
What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml: <extension name="setup_media" continue="true"> <condition field="${sip_nat_detected}" expression="true"> <action application="set" data="proxy_media=true" /> <action application="set" data="bypass_media=false" /> <anti-action application="set" data="proxy_media=false" /> <anti-action application="set" data="bypass_media=true" /> </condition> </extension>
I was using revision 15586. ----------------------------------------
From: brian@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
What rev exactly?
/b
On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml:
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141665/direct/01/
This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. Mike On Nov 23, 2009, at 11:33 PM, John Platts wrote:
I was using revision 15586.
----------------------------------------
From: brian@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
What rev exactly?
/b
On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml:
I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH. ________________________________
From: mike@jerris.com Date: Tue, 24 Nov 2009 03:39:16 -0500 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again.
Mike
On Nov 23, 2009, at 11:33 PM, John Platts wrote:
I was using revision 15586.
---------------------------------------- From: brian@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
What rev exactly?
/b
On Nov 23, 2009, at 6:19 PM, John Platts wrote:
I actually checked out the latest version of FreeSWITCH in the SVN repository.
I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml:
_________________________________________________________________ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/
Are you sure you did a make current? and can you outline the issue in more detail? /b On Nov 24, 2009, at 3:28 PM, John Platts wrote:
I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH.
participants (3)
-
Brian West
-
John Platts
-
Michael Jerris