Hello, Endpoints involved: Cisco UCM FreeSWITCH (latest 1.6 GIT) Call flow is rather simple: <param name="enable-3pcc" value="proxy”/> <param name="inbound-late-negotiation" value="true"/> proxy and bypass media commented out Receive <extension name="from_cucm"> <condition field="network_addr" expression="^10\.0\.64\.6[1-6]$"> <action application="pre_answer"/> <action application="transfer" data="${destination_number} XML internal"/> </condition> </extension> And in Internal plan route to carrier: <extension name="mts_tascom"> <condition field="destination_number" expression="^(17[0-5]\d\d)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/mts_tascom-1/$1" /> </condition> </extension> recv 1221 bytes from udp/[10.0.64.63]:6060 at 20:34:55.093189: ------------------------------------------------------------------------ INVITE sip:17000@10.0.64.100:6060 SIP/2.0 Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300821e56a448 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100> Date: Wed, 16 Sep 2015 17:34:55 GMT Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a@10.0.64.63 Supported: 100rel,timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CP7942G/9.3.1 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence, kpml Supported: X-cisco-srtp-fallback,X-cisco-original-called Call-Info: <sip:10.0.64.63:6060>;method="NOTIFY;Event=telephone-event;Duration=500" Cisco-Guid: 1025134336-0000065536-0000139308-1061158922 P-Charging-Vector: icid-value="3D1A4F00000100000002202B3F40000A";icid-generated-at=msk-vo-s-ucm3;orig-ioi="IMS Inter Operator Identification" Session-Expires: 1800 P-Asserted-Identity: "IT, ????? ??????" <sip:12550@10.0.64.63> Remote-Party-ID: "IT, ????? ??????" <sip:12550@10.0.64.63>;party=calling;screen=yes;privacy=off Contact: <sip:12550@10.0.64.63:6060> Max-Forwards: 69 Content-Length: 0 ------------------------------------------------------------------------ send 400 bytes to udp/[10.0.64.63]:6060 at 20:34:55.093460: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300821e56a448 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100> Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a@10.0.64.63 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Content-Length: 0 ------------------------------------------------------------------------ send 1676 bytes to udp/[10.0.64.63]:6060 at 20:34:55.095896: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK300821e56a448 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100>;tag=9NZXy4BKSB4cm Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a@10.0.64.63 CSeq: 101 INVITE Contact: <sip:mod_sofia@10.0.64.100:6060> RSeq: 2100659333 User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Require: 100rel Supported: precondition, 100rel, timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 901 v=0 o=FreeSWITCH 1442403045 1442403046 IN IP4 10.0.64.100 s=FreeSWITCH c=IN IP4 10.0.64.100 t=0 0 a=msid-semantic: WMS no6JiXqEFi18v7d8DbGarc1t7srxqe0E m=audio 21850 RTP/SAVPF 18 9 8 0 3 101 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fingerprint:sha-256 09:0E:1F:D3:0D:FD:F9:54:AA:FF:AD:7F:CC:0A:85:94:D6:16:38:9F:94:43:3F:78:AC:E8:FC:8B:2B:0C:83:CA a=setup:actpass a=rtcp-mux a=rtcp:21850 IN IP4 10.0.64.100 a=ssrc:1443087231 cname:sXdALt5gCXfXVjVo a=ssrc:1443087231 msid:no6JiXqEFi18v7d8DbGarc1t7srxqe0E a0 a=ssrc:1443087231 mslabel:no6JiXqEFi18v7d8DbGarc1t7srxqe0E a=ssrc:1443087231 label:no6JiXqEFi18v7d8DbGarc1t7srxqe0Ea0 a=ice-ufrag:ksoCugWrzdPPiGkB a=ice-pwd:un25HYb9LhXxqILA0Z2l6ODw a=candidate:9999950793 1 udp 659136 10.0.64.100 21850 typ host generation 0 a=ptime:20 ------------------------------------------------------------------------ recv 888 bytes from udp/[10.0.64.63]:6060 at 20:34:55.101106: ------------------------------------------------------------------------ PRACK sip:mod_sofia@10.0.64.100:6060 SIP/2.0 Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK3008370972459 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100>;tag=9NZXy4BKSB4cm Date: Wed, 16 Sep 2015 17:34:55 GMT Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a@10.0.64.63 CSeq: 102 PRACK RAck: 2100659333 101 INVITE Allow-Events: presence, kpml Max-Forwards: 70 Content-Type: application/sdp Content-Length: 376 v=0 o=CiscoSystemsCCM-SIP 357909 1 IN IP4 10.0.64.63 s=SIP Call c=IN IP4 0.0.0.0 t=0 0 m=audio 21850 RTP/AVP 9 0 8 3 18 101 a=rtcp-mux a=rtpmap:9 G722/8000 a=ptime:20 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=inactive a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 617 bytes to udp/[10.0.64.63]:6060 at 20:34:55.101291: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.64.63:6060;branch=z9hG4bK3008370972459 From: "IT, ????? ??????" <sip:12550@10.0.64.63>;tag=357909~27154efa-6325-45a2-9e47-67e5d9302ebc-249342896 To: <sip:17000@10.0.64.100>;tag=9NZXy4BKSB4cm Call-ID: 3d1a4f00-5f91a83f-24684-3f40000a@10.0.64.63 CSeq: 102 PRACK Contact: <sip:mod_sofia@10.0.64.100:6060> User-Agent: FreeSWITCH-mod_sofia/1.6.0+git~20150903T203652Z~6762f14140~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, PRACK, NOTIFY Supported: precondition, 100rel, timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ On 24 Sep 2015, at 19:45, Brian West <brian@freeswitch.org> wrote:
What is the call flow and devices involved? You shouldn't be getting the webrtc SDP if your bridge line and settings are correct, can you elaborate? Also that second part looks like a bug those should go on JIRA.
-- Serge S. Yuriev