<div>I am debugging it again with these settings</div>
<div> </div>
<div>rtp payload-type nse 100<br> rtp payload-type nte 101</div>
<div> </div>
<div>I am having a really hard time from the telco getting what settings I should use or even what settings to use or even what system they are using in the back end. Is there anyway to tell from the debug?</div>
<div> </div>
<div><strong>I get the following below</strong><br></div>
<div>Sent:<br>INVITE <a href="http://sip:18774675464@64.154.41.200:5060">sip:18774675464@64.154.41.200:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>Remote-Party-ID: <<a href="mailto:sip%3A6782282221@173.14.220.57">sip:6782282221@173.14.220.57</a>>;party=calling;screen=yes;privacy=off<br>
From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">sip:18774675464@64.154.41.200</a>><br>Date: Tue, 27 Oct 2009 12:34:09 GMT<br>
Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>Supported: 100rel,timer,resource-priority,replaces,sdp-anat<br>Min-SE: 1800<br>Cisco-Guid: 2157240972-3604177326-402682881-167847941<br>
User-Agent: Cisco-SIPGateway/IOS-12.x<br>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER<br>CSeq: 102 INVITE<br>Max-Forwards: 70<br>Timestamp: 1256646849<br>Contact: <<a href="http://sip:6782282221@173.14.220.57:5060">sip:6782282221@173.14.220.57:5060</a>><br>
Expires: 180<br>Allow-Events: telephone-event<br>Proxy-Authorization: Digest username="1648245954",realm="64.154.41.110",uri="<a href="http://sip:18774675464@64.154.41.200:5060">sip:18774675464@64.154.41.200:5060</a>",response="ab63d4755ff4182631ad2db0f9ed0e44",nonce="12901115532:303fa5d884d6d0a5a0328a838545395b",algorithm=MD5<br>
Content-Type: application/sdp<br>Content-Disposition: session;handling=required<br>Content-Length: 250</div>
<div>v=0<br>o=CiscoSystemsSIP-GW-UserAgent 7043 4703 IN IP4 173.14.220.57<br>s=SIP Call<br>c=IN IP4 173.14.220.57<br>t=0 0<br>m=audio 16462 RTP/AVP 0 100<br>c=IN IP4 173.14.220.57<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:100 telephone-event/8000<br>
a=fmtp:100 0-15<br>a=ptime:20</div>
<div>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog<br>*Oct 27 12:34:09.332: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">sip:18774675464@64.154.41.200</a>><br>
Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>CSeq: 102 INVITE<br>Content-Length: 0</div>
<div><br>*Oct 27 12:34:09.332: //846/8094E28C1800/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX<br>*Oct 27 12:34:09.332: //846/8094E28C1800/SIP/State/sipSPIChangeState: 0x4A357FCC : State change from (STATE_SENT_INVITE, SUBSTATE_NONE) to (STATE_RECD_PROCEEDING, SUBSTATE_PROCEEDING_PROCEEDING)<br>
*Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpIPv4SocketReads: Msg enqueued for SPI with IP addr: [64.154.41.200]:5060<br>*Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 for event 1<br>
*Oct 27 12:34:10.832: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_new_msg_preprocessor: Checking Invite Dialog<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received:<br>SIP/2.0 183 Session Progress<br>To: <<a href="mailto:sip%3A18774675464@64.154.41.200">sip:18774675464@64.154.41.200</a>>;tag=3465630735-938664<br>From: <<a href="mailto:sip%3A6782282221@sip.talkinip.net">sip:6782282221@sip.talkinip.net</a>>;tag=2EDA9C8-25D6<br>
Contact: <<a href="http://sip:18774675464@64.154.41.200:5060">sip:18774675464@64.154.41.200:5060</a>><br>Call-ID: <a href="mailto:DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57">DB9895B8-C22B11DE-801EC992-790F56B7@173.14.220.57</a><br>
CSeq: 102 INVITE<br>Content-Type: application/sdp<br>Via: SIP/2.0/UDP 173.14.220.57:5060;branch=z9hG4bK4A18DE<br>Content-Length: 146</div>
<div>v=0<br>o=msx71 490 6110 IN IP4 64.154.41.200<br>s=sip call<br>c=IN IP4 64.154.41.101<br>t=0 0<br>m=audio 45846 RTP/AVP 0<br>a=ptime:20<br>a=rtpmap:0 PCMU/8000</div>
<div>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckResponse: INVITE response with no RSEQ - disable IS_REL1XX<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sipSPIGetContentGTD: No GTD found in inbound container<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoMediaNegotiation: Number of m-lines = 1<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.14.220.57<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation: Codec (g711ulaw) Negotiation Successful on Static Payload for m-line 1<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoPtimeNegotiation: One ptime attribute found - value:20<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/convert_ptime_to_codec_bytes: Values :Codec: g711ulaw ptime :20, codecbytes: 160<br>
*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIDoPtimeNegotiation: Offered ptime:20, Negotiated ptime:20 Negotiated codec bytes: 160 for codec g711ulaw<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: m-line index 1<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPICheckDynPayloadUse: Dynamic payload(100) could not be reserved.<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoDTMFRelayNegotiation: Case of full named event(NE) match in fmtp list of events.<br>*Oct 27 12:34:10.836: //-1/xxxxxxxxxxxx/SIP/Info/sip_sdp_get_modem_relay_cap_params: NSE payload from X-cap = 0<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sip_select_modem_relay_params: X-tmr not present in SDP. Disable modem relay<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIGetSDPDirectionAttribute: No direction attribute present or multiple direction attributes that can't be handled for m-line:1 and num-a-lines:0<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Info/sipSPIDoAudioNegotiation: Codec negotiation successful for media line 1<br> payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=rtp-nte<br> stream_type=voice+dtmf (1), dest_ip_address=64.154.41.101, dest_port=45846<br>
*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/State/sipSPIChangeStreamState: Stream (callid = -1) State changed from (STREAM_DEAD) to (STREAM_ADDING)<br>*Oct 27 12:34:10.836: //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:<br>
Preferred Codec : g711ulaw, bytes :160<br> Preferred DTMF relay : rtp-nte<br> Preferred NTE payload : 100<br> Early Media : No<br> Delayed Media : No<br> Bridge Done : No<br>
New Media : No<br> DSP DNLD Reqd : No</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/resolve_media_ip_address_to_bind: Media already bound, use existing source_media_ip_addr<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPISetMediaSrcAddr: Media src addr for stream 1 = 173.14.220.57<br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br> callId 846 peer 845 flags 0x200005 state STATE_RECD_PROCEEDING<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>
CallID 846, sdp 0x497E29C0 channels 0x4A35926C<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:<br> callId 846 size 240 ptr 0x4A170B28)<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>
Hndl ptype 0 mline 1<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Selecting codec g711ulaw<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/codec_found:<br>Codec to be matched: 5<br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: ADD AUDIO CODEC 5</div>
<div>*Oct 27 12:34:10.840: //-1/xxxxxxxxxxxx/SIP/Info/convert_codec_bytes_to_ptime: Values :Codec: g711ulaw codecbytes :160, ptime: 20<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Media negotiation done: stream->negotiated_ptime=20,stream->negotiated_codec_bytes=160, coverted ptime=20 stream->mline_index=1, media_ndx=1<br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo:<br>Adding codec 5 ptype 0 time 20, bytes 160 as channel 0 mline 1 ss 1 <a href="http://64.154.41.101:45846">64.154.41.101:45846</a><br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = 5</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_copy_sdp_to_channelInfo: Copy sdp to channel- AFTER CODEC FILTERING: ccb->pld.ipip_caps.codecInfo[channel_ndx].codec = -1</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br> callId 846 flags 0x100 state STATE_RECD_PROCEEDING<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer:<br>
Report initial call media<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: ccb->flags 0x400018, ccb->pld.flags_ipip 0x200005</div>
<div>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/copy_channels:<br> callId 846 size 240 ptr 0x4DEC000C)<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/ccsip_update_srtp_caps: 5030: Posting Remote SRTP caps to other callleg.<br>
*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Info/sipSPI_ipip_report_media_to_peer: do cc_api_caps_ind()<br>*Oct 27 12:34:10.840: //846/8094E28C1800/SIP/Media/sipSPIUpdCallWithSdpInfo:<br> Stream type : voice+dtmf<br>
Media line : 1<br> State : STREAM_ADDING (2)<br> Stream address type : 1<br> Callid : 846<br> Negotiated Codec : g711ulaw, bytes :160<br>
Nego. Codec payload : 0 (tx), 0 (rx)<br> Negotiated DTMF relay : rtp-nte<br> Negotiated NTE payload : 100 (tx), 100 (rx)<br> Negotiated CN payload : 0<br> Media Srce Addr/Port : [173.14.220.57]:16462<br>
Media Dest Addr/Port : [64.154.41.101]:45846</div>
<div><br> </div>
<div> </div>
<div> </div>