-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, correction: 6. OK with SDP (T.38) from freeswitch back to fax A is wrong. It should be 6. OK with SDP (voice) from freeswitch back to fax A Helmut Kuper schrieb:
Hello again,
Update:
my environment looks like this:
kapanga soft phone -> openser -> freeswitch -> SBC-> PSTN-Gateway
T.38 call setup seems to work like this:
1. INVITE with SDP (voice) to PSTN Gateway (fax B) 2. Session in Progress with SDP (voice) back to fax A 3. OK with SDP (voice) back to fax A 4. Faxdetection in PSTN Gateway 5. re-INVITE with SDP (T.38) from PSTN-Gateway to fax A 6. fax exchange
With freeswitch it looks like this:
1. INVITE with SDP (voice) to PSTN Gateway terminated by freeswitch 2. INVITE with SDP (voice) from freeswitch to PSTN-Gateway 3. Session in Progress with SDP (voice) back to freeswitch 4. Session in Progress with SDP (voice) from freeswitch back to fax A 5. OK with SDP (voice) back to freeswitch 6. OK with SDP (T.38) from freeswitch back to fax A 7. Voice exchange 8. Faxdetection in PSTN Gateway 9. re-INVITE with SDP (T.38) from PSTN-Gateway to freeswitch 10. Codec Error in freeswitch/Sofia
It's seems to me that freeswitch should not touch the SDP in this case, but should send in transparently from pstn gateway to fax A.
I looked into freeswutch's online docu and found this, which could help :
"No Media Mode
No media mode is an SDP Passthough feature that permits to endpoints that can see each other (no funky NAT's) to connect their media sessions directly while FreeSwitch maintains control of the SIP signalling. This is useful if you have two end-points that need to use a codec that is currently not supported in FreeSwitch (video) or if you are using FreeSwitch in a high performance walled garden network and want to minimize the RTP handling FreeSwitch is doing to maximize call traffic.
Before executing the bridge action you must set the "no-media" flag to true. "
I changed my dialplan to this:
----------------- <extension name="PrintVars" continue="true"> <condition field="destination_number" expression="(.*)"> <action application="set" data="no_media=true"/> </condition> </extension>
<extension name="Fax dialout"> <condition field="destination_number" expression="^94(\d{4})$"> <action application="set" data="effective_caller_id_name=032223930016"/> <action application="set" data="effective_caller_id_number=032223930016"/> <action application="set" data="call_timeout=10"/> <action application="info"/> <action application="bridge" data="sofia/gateway/sip01.voice.ewetel.de/04418000$1@sip01.voice.ewetel.de"/> </condition> </extension> --------------------------------
Unfortunately freeswitch still changes the SDP as a full B2BUA.
I tried various values for no_media: true, TRUE, 1 but no change. I even put it directly into "Fax dialout" exttension. It seems freeswitch ignores this setting. At least it is printed out on console by info action.
I attached the log of freswitch's answere to pstn gateway's re-INVITE.
Any ideas what I do wrong ?
regards helmut
----------LOG-----------------
recv 740 bytes from udp/[213.168.198.84]:5060 at 08:08:50.367892: ------------------------------------------------------------------------ INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Max-Forwards: 69 Contact: <sip:044180002799@213.168.198.84:5060;transport=udp> Content-Length: 274 Content-Type: application/sdp
v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy ------------------------------------------------------------------------ tport_deliver(0x8091ed8): msg 0x809a2e0 (740 bytes) from udp/213.168.198.84:5090/sip next=(nil) nta: received INVITE sip:032223930016@212.6.123.118:5090 SIP/2.0 (CSeq 210) nta: canonizing sip:032223930016@212.6.123.118:5090 with contact nta: INVITE (210) going to existing leg nta: timer shortened to 200 ms nua: nua_stack_process_request: entering soa_init_offer_answer(static::0x80c59c8) called soa_set_remote_sdp(static::0x80c59c8, (nil), 0x809444a, 274) called nua: nua_invite_server_respond: entering soa_generate_answer(static::0x80c59c8) called soa_static_offer_answer_action(0x80c59c8, soa_generate_answer): called soa_static(0x80c59c8, soa_generate_answer): upgrade with remote description soa_static(0x80c59c8, soa_generate_answer): marking rejected media soa_sdp_mode_set_is_needed(0x413fec44, 0x80d1e50, ""): called soa_static(0x80c59c8, soa_generate_answer): storing local description soa_activate(static::0x80c59c8, (nil)) called soa_get_local_sdp(static::0x80c59c8, [(nil)], [0x413fed58], [0x413fed54]) called tport_tsend(0x8091ed8) tpn = UDP/213.168.198.84:5060 tport_resolve addrinfo = 213.168.198.84:5060 tport_by_addrinfo(0x8091ed8): not found by name UDP/213.168.198.84:5060 tport_vsend(0x8091ed8): 771 bytes of 771 to UDP/213.168.198.84:5060 tport_vsend returned 771 send 771 bytes to udp/[213.168.198.84]:5060 at 08:08:50.369576: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1 From: <sip:044180002799@sip01.voice.ewetel.de>;tag=ejgpi49boo292 To: <sip:032223930016@sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF Call-ID: edcb6623-afa8-122a-bda0-005056947ed9 CSeq: 210 INVITE Contact: <sip:212.6.123.118:5090> User-Agent: FreeSWITCH(mod_sofia) Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 156
v=0 o=FreeSWITCH 4751724862107259708 3984222328945838171 IN IP4 212.6.123.118 s=FreeSWITCH c=IN IP4 212.6.123.118 t=0 0 a=sendrecv m=image 0 UDPTL 9 ------------------------------------------------------------------------ nta: sent 200 OK for INVITE (210) nua(0x80c4e90): event i_invite 200 OK nua(0x80c4e90): ready call updated: completed received offer, sent answer soa_get_remote_sdp(static::0x80c59c8, [0x413fedd8], [0x413fedd4], [(nil)]) called soa_get_local_sdp(static::0x80c59c8, [0x413fedd0], [0x413fedcc], [(nil)]) called soa_get_params(static::0x80c59c8, ...) called nua(0x80c4e90): event i_state 200 OK nua: nua_event: entering nua: nua_event: entering 2007-07-18 10:08:50 [DEBUG] sofia.c:1028 sofia_handle_sip_i_state() Channel sofia/sip0.en.ewetel.de/044180002799@sip01.voice.ewetel.de entering state [completed] 2007-07-18 10:08:50 [DEBUG] sofia.c:1032 sofia_handle_sip_i_state() Remote SDP: v=0 o=bell 1219258 1219259 IN IP4 213.168.198.84 s=- c=IN IP4 213.168.198.84 t=0 0 m=image 10926 UDPTL t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:336 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy
2007-07-18 10:08:50 [ERR] sofia.c:1218 sofia_handle_sip_i_state() Reinvite Codec Error!
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