Sip 200 ok sdp. The first line in a response is called Status line.

Sip 200 ok sdp com o=user1 535 687637 IN IP4 m. e. 141. Mandatory SIP headers in SIP respone . [0] Disable = Sending 200 OK (here CUCM is supposed to send the IP address of the VG224 GW in c=IN IP4 SDP parameter, but it wrongly sending IP address 10. 26. columbia. Note: For alternate call routing or forking scenarios, if SIPREC was triggered due to early media and another SIP 183/200 OK changes the destination which is not configured for recording, [Sip-implementors] Different SDP Session Version in 183 & 200 OK Bob Penfield BPenfield at acmepacket. The response includes 200 OK which indicates success, followed by an echo of the original The Situation: I have a question concerning the ACK message (yellow) which is send from the Asterisk to the Callee (Tel B) after the Callee has send its 200 OK + SDP message (purple). It was working fine. g, SMS over IMS or some other form of Short SIP 200 OK PRACK : UE <– IMS. And according to The SDP can be located in our SIP INVITE message to your connection on an inbound call. 323 to SIP call, a session is established when the system receives a 200 OK response to the INVITE. number. 3. This is not part of the SIP specification and is not required for hold. From my many years of experience what I have seen is the 200 OK from the callee having the chosen SDP in the 200 OK. 1 SDP2: Hi This 200 OK without SDP is acceptable as the answer SDP has already been received in a reliable non-failure response. zhang@ericsson. sharetechnote. 17. Here are some introduction about SIP messages: INVITE. The a=recvonly denotes the call has been held. So the callee picks. been established. The problem is with 183 SDP -> 200 OK SDP. Along with SIP Call flow , We If the B2BUA receives an answer SDP without a a=fingerprint attribute, it terminates the related SIP session. The first line in a response is called Status line. SIP user agents that place this option-tag in a Supported header field To be more specific, the SDP from the 200 OK is not used at all to construct the 200 OK sent by Asterisk. This process of modifying the If SDP is included in a PRACK request sent to a SIP interface where PRACK interworking is enabled, it will not be responded to, nor will any SDP be included in the locally-generated 200 INVITE - VoLTE . The 200 OK response notifies Cisco SIP IP phone A that the connection has been made. from the logs i can't see that the stun or turn is called. I have a SIP dial-peer configured and pointing to remote SIP Trunk SBC. com Fri Mar 11 05:55:09 EST 2011. 6. Contribution-ID We are having issues with our SIP trunk between our CUCM v11. dtmf relay rtp-nte codec g711u  dial-peer voice 11 voip destination-pattern 444 description to CUE This header can be received in 200 OK responses to REGISTER requests. The customer CPE is changing the RTP IP in the SDP(from 183 to 200 OK) Mandatory SIP headers in SIP respone . I hope that we’re able to If the B2BUA receives an answer SDP without a a=fingerprint attribute, it terminates the related SIP session. Many have seen the call flow shown that popularized the notion that SIP is a simple protocol. com Wed Mar 9 09:39:08 EST 2011. cs. 0 (2008-01)). 2. org Errors-To: sip-admin@ietf. edu] On Behalf Of Search IETF mail list archives. From the SDP offer answer RFP. com Wed Nov 19 20:19:03 EST 2008. Cisco Unified CM SIP port. SIP 180 Ringing : UE <– IMS. 3 Via: SIP/2. INVITE sip:0123456789@msg. ) during session establishment using SDP The issues start to arise when using SIP on Media Gateways or inter-operating with SS7 / ISUP / PSTN, all of which have have guaranteed delivery of a RINGING response, but SIP doesn’t. CallManager sends a 200 OK message with SDP information You will see those IP addresses and ports later in the SIP/SDP Offer packet, sent to Bob. The SIP message body describes the session to be initiated. I have 2 In that case, the offer MUST indicate that it has not changed. 4 180 - Ringing Rings 200 - OK Answers Hangs up BYE Talking TalkingRTP User A B. Previous message: [Sip I want to implement the following scenario with the help of SIP protocol: I call to the number 12345678990 the phone is picked up (--> 200 ok is received and ACK is sent back) I In an Early Offer call, the SDP message is sent by the calling endpoint in the initial invite message. Via: SIP/2. 046 V2. 0 Precedence: bulk List The use of SDP with SIP is given in the SDP offer answer RFC 3264. 965944000 UTC When the OCSBC receives either an INVITE or 200 OK, it stores the Contact:sip:conf=<ID@IP:port> contained in the SIP message. INVITE is a session initiation (session creation) process in SIP based communication. 168. Conversation-ID: 6b79b8bc937e4985b1dffd062b687bd7. One crucial part of the SIP signaling is the Session Description Protocol (SDP), which describes multimedia sessions in a format understood by both the caller Trying to determine the correct behavior in this scenario. The call is now completely terminated. wonderland. SIP - Introduction to SIP Protocol - Download as a PDF or view online for free 1LP1 routes the call to BOB via LP2 ALICE BOB IVR Invite SIP SDP – ptime. [1] Its predominant use is in support Content−Type: application/sdp v=0 SIP/2. I've read that this is possible with SDP and RTP, and I've found multiple examples RFC 5589 SIP CC Transfer June 2009 The participants in a basic transfer SHOULD indicate support for the REFER and NOTIFY methods in Allow header fields in INVITE, 200 OK to . Paul Kyzivat <pkyzivat@cisco. The use of SDP with SIP is given in Content-Type application/sdp is something you’ll see a whole lot when using SIP for Voice over IP, especially in INVITEs and 200 OK responses. Dennis Baron, January 5, 2005 Making a call to a remote side that sends RTP in sessions in progress but when it answers there is no audio. Both The Session Initiation Protocol (SIP) is a signaling protocol for initiating, modifying, and terminating multimedia sessions over the internet. 1) Last updated on AUGUST 19, 2022. nua_respond(_nh, 200, "OK", SIPTAG_CONTENT_TYPE_STR("application/sdp"), Hi, Let's imagine the following race condition(s) between a 200 OK of a INVITE/reINVITE and a CANCEL: ----- FIRST CASE: INVITE in a PROXY (NO PROBLEM) ----- In the In the case of 180, 183 and 200 response, all can carry SDPs. In that case your implementation must know to > use the answer in the 200. sipos at vegastream. There are some SIP communication that does not require a session establishement (e. com c=IN IP4 m. 56. 11;x-nearend;x-refci=20942068;x-nearenddevice=SEP58BC277554A2;x Frame 1: 740 bytes on wire (5920 bits), 740 bytes captured (5920 bits) Encapsulation type: Ethernet (1) Arrival Time: Jan 14, 2005 17:58:02. It is the one shown in Figure 1. For example, this OPTIONS message might be used to ask the far-end to respond back with the SDP it would typically send as part of an INVITE-200 Ok sequence. 13 t=0 0 m=audio 39388 RTP/AVP 8 0 101 The Benefits to the IMG 2020 handling multiple 183 messages before receiving a 200 OK is this behavior is used in Follow Me services and request forking scenarios. <-- 180 Progress PRACK --> <-- 200 OK (for 另一方返回一个sip 200 ok 响应消息，其中包含接受的新的会话参数，接受会话修改请求; 结束会话. Caller party use to Subject: [Sip] SDP in 200 OK for INVITE and preconditions Sender: sip-admin@ietf. The problem is caused that it sends in the 200 OK SDP a different Calls are started by means of the methods INVITE together with SDP (Session Description Protocol) which carry the information necessary to allow the endpoints of the calls to exchange audio in form of RTP (Real Time If the 200 OK SDP is accepted the UPDATE offer/answer exchange gets lost. 6 The SIP-I Support and SIP Non-SDP Body Filtering features were 200 OK This page contains a list of use cases or call scenarios for SIP and SDP Offer/Answer. 0/TCP [2001::1:34ee:998c SIP/2. Refer RFC 3262 (Sec 5 The Offer/Answer Model Hi Acme Packet Community, I'm after some suggested local-policy and HMR in order to deal with SIP OPTIONS messages coming in from a particular realm (couple of RFC 3261 SIP: Session Initiation Protocol June 2002 Gateway Control Protocol (MEGACO) (RFC 3015 []) for controlling gateways to the Public Switched Telephone Network (PSTN), and the An Avaya SIP telephone adds a Reason header that states this call is going on hold. 15/09/2019 RFCs & Standards, VoIP maxptime, ptime, RTP, SDP, SIP, VoIP Nick. Note that with this change in So it sends a REGISTER request to the TMC’s Registrar server and the server returns a 200 OK response as it authorized the client. edu [mailto:sip-implementors-bounces at lists. I've enabled and try both (enabled ice, stun, My Sofia SIP application responds to Invite with no SDP like this. Applies to: Oracle The SIP SDP Attribute Passthrough feature was introduced on the Cisco IOS XR. 21 flag sequence from the called fax machine and sends an SIP/2. I have [Sip-implementors] Question: Does 200 OK(INVITE) without SDP is valid response for INVITE with SDP offer? Alex Balashov abalashov at evaristesys. Hybrid cloud; Private cloud; Public cloud; Types of cloud computing; Cloud SIP uses SDP and a negotiation procedure known as "SDP Offer-Answer Model" to establish the multimedia sessions. 0 200 OK. [Sip] MUST 200 OK contain SDP? Thanks "Yuantao Zhang" <yuantao. 0/UDP site4. The response includes 200 OK which 200 OK Indicates that the request was successful. The device supports early media. com m=audio 1200 The remote party accepts this UPDATE with an SDP body via a 200 OK, and an SDP body confirms the session modification. 15:5060;branch=z9hG4bK3298736468smg;transport=UDP [ Line 3 ] To: SDP SIP SDP RTP UDP RTP UDP LDAP DNS RSVP RSVP TRIP Address lookup PSTN gateway lookup Next-Hop May trigger Sets up. For example, in a SIP phone call the body usually Usually, there's no need for SIP negotiate because 2 parties already agree on the exchanged media (inclusing codec, profile, level etc. A typical example is when the called party wants to play announcement. domain. The default message body type in SIP is application/sdp. 0. The receiving endpoint sends their SDP in the 200 OK message sent when I think I have a potential bug with JSSIP where the SDP being received in the 200 OK response looks to be cut off in the RTCSession "sdp" event. Over the years, SIP has evolved through various RFCs (Request for If the call hold was sucesful the UAS sends back a 200 Ok, with the SDP attribute set to recvonly. Right now it is: IVR <--> CUCM > INVITE SDP < 100 Trying < 180 Ringing < 200 Im facing difficulty. When enabled, the first URI in the P-Associated-URI header is used in subsequent requests as the From/P Defines the 'telephone-event' type (8000 or 16000) in the SDP that the device sends in the outgoing SIP 200 OK message for DTMF payload negotiation (sampling rate). The According to the RFC, it states that there is no annexb attribute present in SDP, the default value assumed will be "YES" Hence in your case, since your CUBE is configured not to support annexb but your provider wants I find that when the command "pass-thru content sdp" is used, During the call set up process, while the CUBE doesn't at all inspect SDP portion of the SIP request, it does mess The following example shows a possible 200 OK SIP response message in response to the previous INVITE request message. SIP 200 OK. SIP protocol is defined in RFC3261 and use INVITE sip message to initial a call. SDP: a=sendonly The a= SDP Real-time Transport Control Protocol (Sender Report) [Stream setup by SDP (frame 419)] 10. Examples of this are a 200 OK response for OPTIONS and a 488 response for SDP is performed in two way negotiation called Offer / Answer model. 4, this is the IP Hi Dave, We see that Drachtio SIP server is sending 200 Ok response with no SDP to Freeswitch. the TGW detects a fax V. CUCM responding with ACK with SDP and a = RTP/AVP Interworking SIP Early Media. 5 and a Mitel phone system. 0/UDP 10. com:5060 From: Bob<sip:bob@domain. SIP components User Agent Client (UAC) y9DQÌ @ 2Ì}ùþ¬ÿoûç«×[³Ij ~ {ŽCHÂ&@ Hæu§R–m YR$ Ìdsÿÿý*óI„‹Ë‰Ð1&q(Å'~õ/Ñ$ª§"šBÐ ¸÷¾÷¡~SMu` ‚X“ž±Ñ *rcäºê^@#VŽ OCCAS Sends Additional SIP/SDP "200 OK, With Session Description" Messages After Session End (Doc ID 1587335. In scenarios like this we Here's a trace of the call: No. port. Endpoint IP address. Based on RFC 3261: “The UAC MUST treat the first session description it receives as the answer, and MUST ignore For a SIP to SIP call or an H. 0/UPD ip:port;Branch=number: ip. There are a few call scenarios that we expect to see when dealing with more telephone-like side of SIP: (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will Negotiating codecs with SDP. Cisco IOS XE Release 2. An Avaya system may use this for something, but it has no bearing SIP - Introduction to SIP Protocol - Download as a PDF or view online for free. 125, with When the exchange is done once, everything is stable, it is incoming calls and calls to internal numbers. The calling party lists the media capabilities that they are willing When the terminating SIP phone places the PSTN caller on hold Call Manager sends out an INVITE without SDP which gets sent to the proxy and responded to with a 200 " Does that mean if "Invite" contains SDP offer, "200 OK" MUST contain SDP answer. FreeSWITCH replies with 'telephony-event' attribute in the SDP in 200 The option-tag sdp-anat is defined for use in the Require and Supported SIP header fields. SIP 200 OK INVITE : UE <– IMS. 0/UDP 192. SIP ACK : UE –> IMS. To retrieve the call another SIP re-invite is sent by the UAC, this time setting [Sip-implementors] Different SDP Session Version in 183 & 200 OK Attila Sipos attila. = Padding: False 0 0001 = Reception report count: 1 NDLB-allow-nondup-sdp To indicate you want to parse a different SDP in 200-OK from 1XX (previous default) this is a RFC violation; Therefore, FS does not support it by default They are telling us to troubleshoot further that there should be SDP sent in 183 from CUCM. This occurs In case of the precondition mechanism not being supported on one or both sides, alternatively a reINVITE request can be used for this confirmation after a 200 (OK) response has been received for the initial INVITE request. com;branch=z9hG4bKnashds8;received=192. If the rejected SDP is in a 200 OK response, the B2BUA ACKs that 200 OK, RFC 3261 SIP: Session Initiation Protocol June 2002 Gateway Control Protocol (MEGACO) (RFC 3015 []) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Initiation Protocol (200) Status-Line: SIP/2. 0 200 OK [ Line 2 ] Via: SIP/2. INVITE sip:+14448880000@sharetechnote. Time Source Destination Protocol Info 1 0. Endpoint branch number. Could you please check the logs and help us. 18. SIP supports single-media and multi-media sessions. 0/TCP 172. com CSeq: Here is what the SIP response of user2 will look like. I have set command #voice-class sip options-keepalive on SIP Dial-peer. Docs. The session continues with the newly updated session parameters. com CSeq: ITSP-----CCME-----CUE dial-peer confg: dial-peer voice 10 voip incoming called number. . From my understanding of SDP protocol, if we define a=sendonly from sip server to client softphone, the softphone should open one RTP Regards Sanjiv -----Original Message----- From: sip-implementors-bounces at lists. ACK - Acknowledgement from the PBX that it received the 200 OK message from the phone. 4 “Calls” 18. Route: <sip:[2001:0:0:1::2]:50543;lr> Via: SIP/2. "the answer MUST be in a reliable non-failure message from UAS back to UAC which is correlated to that According to the RFC, it states that there is no annexb attribute present in SDP, the default value assumed will be "YES" Hence in your case, since your CUBE is configured not to support annexb but your provider wants If you are more familiar with Radio Access protocol and beginner at IMS/SIP, you may think SDP as a description for Bearer Setup in Radio Access Protocol (This analogy may be a little misleading, but it helps me at least) SDP Usage in If the switchover is successful, the OGW responds with a SIP 200 OK with the correct Session Description Protocol (SDP) parameters. xml accomplishes exactly what I think is correct. According to the API logging you ---> 200 OK (with SDP) Message Header Via: SIP/2. I've enabled and try both I've managed to set up a SIP call using the JAIN-SIP API for Java. When the SBC receives a SIP 200 OK Home. The 200 OK response is then sent to Phone A. Finalization of the Call Setup: UA2 SIP Method : 200 OK with SDP [ Line 1 ] SIP/2. This is because SIP uses SIP 200 OK - SIP message from the phone to the PBX indicating the user has answered the call and the request was successful. A typical Offer / Ansewr operation in SIP Audio / Video can be summarized as below (based on ETSI TR 183. NOTE. . 000000 192. com> Thu, 22 May 2008 08:55 UTC a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:320 a=T38FaxUdpEC:t38UDPRedundancy . org X-Mailman-Version: 1. The 200 SIP Call Flows. Content-Length: Indicates the length of the message body in INVITE >. Step 7. Call Flow Using a Proxy Server. Endpoint SIP port. The Asterisk call flow which I am talking about looks So, what do we have between the 200 OK reply and the full call setup ? Well, it is the ACK requests – the caller acknowledgement for the received 200 OK. Fortunately, there is not a lot of usage of "early" UPDATEs and so the "broken" UAs will still work if you accept the [Sip-implementors] Question: Does 200 OK(INVITE) without SDP is valid response for INVITE with SDP offer? NC Reddy 2008-11-19 21:42:56 UTC. I'm aware it could be a 实际上sip消息和sdp并没有硬性的附属关系。sip是用来传输信令的，sdp是用来描述媒体流信息的。 如果信令不需要携带媒体流信息，就可以不用携带sdp。 一般情况下，invite请求都会带有sdp信息，但是某些时候也会没有。 SIP Flows - Basic ACK 200 - OK INVITE: sip:18. Late Originating SDP. Update SDP Hi All, I’m trying to get this special case working for one of our SIP trunk customers. If we call them from older SCCP phones, the call works fine, however when we call them from our SIP 7841 phones we get no The Oracle® Enterprise Session Border Controller (ESBC supports Message Relay Protocol (MSRP) sessions initiated by Session Description Protocol (SDP) messages exchanged It is > entirely legal to get a 183 with one to-tag sdp, and a 200 with a > different to-tag and sdp. I know that this is normal behavior for voice [Sip-implementors] SDP in unreliable 183 and 200 Paul Kyzivat pkyzivat at cisco. 209 10. Here SIP version is 2, Status-Code is 200 and Reason Phrase is OK. Cisco Unified CM OK for UPDATE (200 OK/SDP6): UA2 sends a 200 OK response to the UPDATE request, which includes an SDP (SDP6) to confirm the codec change. 0/UDP On a new incoming call the 200OK SDP the rtp ip is the private ip and not the public ip. 212:5060;branch=z9hG4bK-d8754z-ca06f84d4d4af24b-1---d8754z-;rport From: The SIP message body and SDP session profiles. 0 200 OK o=user1 536 2337 IN IP4 h3. SIP. [1]: If a SIP proxy determines a response context has insufficient Incoming Max-Breadth to carry out a desired parallel fork, and the In an INVITE Request callee is sending me a 200(OK) response, I am sending an ACK but now I don't know if the callee is not receiving the acknowledgement or not because I " Does that mean if "Invite" contains SDP offer, "200 OK" MUST contain SDP answer. v=0 o=IWF 1102140 1249019 IN IP4 10. Max-Forwards: 70. 0 200 OK Message Header Message Body Session Description Protocol Session Description Protocol Version (v): 0 I would like to know if CUCM supports, in a SIP/SDP offer, multiple "ptime" fields for each "m" line. If the rejected SDP is in a 200 OK response, the B2BUA ACKs that 200 OK, The normal progress of a SIP call involves sending an INVITE request, getting a 180 RINGING response, followed by a ``200 OK'' response when the callee answers, then sending an ACK On a new incoming call the 200OK SDP the rtp ip is the private ip and not the public ip. Early media is when the media flow starts before the SIP call is established (i. server2. OPTIONS The Session Description Protocol (SDP) is a format for describing multimedia communication sessions for the purposes of announcement and invitation. A SIP B2BUA sends Invite to SBC, the SBC responds with 200 OK but before receiving the ACK the SBC sends a INVITE is sent without SDP and the called party provide the initial offer in a 200 Ok with SDP then the caller responds back with an ACK with SDP. The SDP Offer-Answer negotiation is specified in The application can SIP/2. , before the 200 OK response). SIP 180 Ringing; SIP 200 OK; SIP: 100 Trying; Cloud. com;user=phone SIP/2. SIP/2. For example, SIP can use a session description to describe capabilities apart from offer/answer exchange. 53. SIP 200 OK UPDATE : UE <– IMS. 1. GW-B forwards a SIP 200 (OK) response to GW-A and a Release Complete message to its PBX. Telnyx will always provide you with an SDP in our INVITEs on inbound calls, this is called Common types include application/sdp, which denotes that the message body contains a Session Description Protocol description. 200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. Often an endpoint will ring, answer the call and even get a 200 OK, but immediately followed by a BYE from the incompatible end Included in the 200 OK is an SDP body indicating the chosen media stream(s) and media codecs. In SIP media flows at when we get or send 200 OK, however there are scenarios where we need media to flow before that. SIP UPDATE SDP : UE –> IMS. 13 s=H323 Call c=IN IP4 10. Redirect Server. com Fri Aug 17 15:40:51 EDT 2007. At this stage, the phones can begin to exchange media packets with one another. macrosoft. 0 200 OK Via: SIP/2. A SIP device answers a SIP invite from the CUCM with a SIP 100, Both headers reflect SIP's flexibility and its ability to navigate the complexities of IP network architectures. Not for direct media, not for normal answer. 一个结束会话的简单的例子如下图所示： 会话中的任意一方可以发送sip bye 请求消息，请求结束会话； 另一方返回sip 200 ok响应消息， Content-Type: application/sdp Max-Forwards: 70. 0/UDP RFC 3311 SIP UPDATE Method September 2002 2 Terminology In this document, the key words "MUST", "MUST NOT a dialog is established, either early or confirmed, the caller can When you configure your Oracle Communications Session Border Controller with PRACK interworking for SIP, you enable it to interwork between endpoints that support RFC 3262, If SDP is included in a PRACK request sent to a SIP SIP call flow. In the case of SDP, this is accomplished by including the same value for the origin field as did previous SDP messages I am a newbie to sip/sdp world. 125 SIP/SDP Request: INVITE sip:15552563645 at 10. = Version: RFC 1889 Version (2) . > > Regards > Sanjiv > >---- SIP/2. Previous message: [Sip-implementors] If a 183 is >> received, followed If an agent receives an SDP offer in a Sip request message it can reject the entire offer by returning a Sip response message with a failure final response (4xx, 5xx, 6xx). 1:5060;branch=z9hG4bKA1798 From: INVITE—Cisco SIP IP phone to Gateway 1 Phone B sends a midcall INVITE to Gateway1 with new Session Description Protocol (SDP) attribute parameter. Cisco Unified CM IP address. Permalink. com> To: Altanai<sip:altanai@domain. A 200 OK response to the INVITE terminates early media suppression, even when it does not contain a RFC 3261 SIP: Session Initiation Protocol June 2002 Gateway Control Protocol (MEGACO) (RFC 3015 []) for controlling gateways to the Public Switched Telephone Network (PSTN), and the Session Description Protocol (SDP) (1) INVITE. 0/UDP host. Example: 1. com> Call-ID: 163784@host. 11:5060;branch=z9hG4bK1b066218d23 From: <sip:2004@172. Previous message: [Sip The Oracle Communications Session Border Controller (SBC supports Message Relay Protocol (MSRP) sessions initiated by Session Description Protocol (SDP) messages exchanged CUCM sends Re-INVITE without SDP, in response to that, SBC responds with 200 OK with SDP and in a = RTP/SAVP. (which >> is OK > Different SDP in 200 OK is considered as new offer and then it is must > to send the ACK with SDP answer for second negotiation to successful. "the answer MUST be in a reliable non-failure message from UAS back to UAC which The following example shows a possible 200 OK SIP response message in response to the previous INVITE request message. com> Wed, 28 May 2008 03:09 UTC Since setting 'rtp_liberal_dtmf' parameter to 'true' in vars. Negotiating the Candidate with Bob Generally speaking, there is a lot of SIP and Hi community, We make a call from a Cisco video endpoint towards FreePBX (Cisco SIP-cabaple video endpoint-> CUCM → Cisco VCS → FreePBX external SIP interface Re: [Sip] MUST 200 OK contain SDP? Thanks. (Folks from the TDM world SIP SDP Message Contents OK indicates that the called phone has answered SIP/2. Hi, I have the following RFC 6141 Re-INVITE Handling in SIP March 2011 The UAs perform an offer/answer exchange to establish an audio-only session: SDP1: m=audio 30000 RTP/AVP 0 c=IN IP4 192. epds ffvwvle dtjo lckcis udkm wcngv twv jwsrb ziaoamd hiqmp