[Freeswitch-users] FreeSwitch and T.38

Wed Jul 18 03:26:05 PDT 2007

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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 at 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 at 212.6.123.118:5090 SIP/2.0
>    Via: SIP/2.0/UDP
> 213.168.198.84:5060;branch=z9hG4bKfnbo0u00907g0bkk43o0sb0000093.1
>    From: <sip:044180002799 at sip01.voice.ewetel.de>;tag=ejgpi49boo292
>    To: <sip:032223930016 at sip01.voice.ewetel.de>;tag=KgBS0cD1D9pZF
>    Call-ID: edcb6623-afa8-122a-bda0-005056947ed9
>    CSeq: 210 INVITE
>    Max-Forwards: 69
>    Contact: <sip:044180002799 at 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 at 212.6.123.118:5090 SIP/2.0 (CSeq 210)
> nta: canonizing sip:032223930016 at 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 at sip01.voice.ewetel.de>;tag=ejgpi49boo292
>    To: <sip:032223930016 at 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 at 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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